Cannot make outgoing calls or receive incoming calls on asterisk 18.23.1

Asterisk Asterisk SIP
1

migrated from sip to pjsip and nothing works. Can’t receive incoming calls, or I can’t place outgoing call. Sip trunk is registered.

<Registration/ServerURI…> <Auth…> <Status…>

reg_us-east-va.sip.flowroute.com/sip:us-east-va.sip.fl auth_reg_us-east-va.sip.flowroute.com Registered (exp. 4s)

Objects found: 1

pjsip.conf

Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
allowoverlap = no

[provider]
insecure = invite

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
local_net=10.0.0.0/24
local_net=127.0.0.1/32
local_net=34.226.36.32/28
external_media_address=xx.xx.xxx.xxx
external_signaling_address=xx.xx.xxx.xxx

[reg_us-east-va.sip.flowroute.com]
type = registration
retry_interval = 20
max_retries = 10
expiration = 120
transport = transport-udp
;transport = transport-udp-nat
outbound_auth = auth_reg_us-east-va.sip.flowroute.com
client_uri = sip:xxxxxxxx@us-east-va.sip.flowroute.com
server_uri = sip:us-east-va.sip.flowroute.com

[auth_reg_us-east-va.sip.flowroute.com]
type = auth
password = AtdhhfGRFfXk
username = 43696067

[hulk]
type = aor
max_contacts = 1

[hulk]
type = auth
username = hulk
password = $$$$$$$$$

[hulk]
type = endpoint
context = phones
allow = alaw,ulaw
auth = hulk
outbound_auth = hulk
aors = hulk

[enigma]
type = aor
max_contacts = 1

[enigma]
type = auth
username = enigma
password = $$$$$$$$$$

[enigma]
type = endpoint
context = phones
allow = alaw,ulaw
auth = enigma
outbound_auth = enigma
aors = enigma

[provider]
type = aor
contact = sip:xxxxxxxx@us-east-va.sip.flowroute.com

[provider]
type = identify
endpoint = provider
match = us-east-va.sip.flowroute.com

[provider]
type = auth
username = provider
password = $$$$$$$$

[provider]
type = endpoint
context = provider
;context = outgoing
allow = ulaw,alaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
from_domain = us-east-va.sip.flowroute.com
auth = provider
outbound_auth = provider
aors = provider

------ Here is the dump of the asterisk log file with logging turned on ------
[2025-06-09 12:02:48.283] Asterisk 18.23.1 built by root @ raspberrypi4 on a aarch64 running Linux on 2024-07-14 16:43:16 UTC
[2025-06-09 12:02:48.283] VERBOSE[3170123] logger.c: Asterisk Queue Logger restarted
[2025-06-09 12:02:59.275] VERBOSE[3170134][C-00000003] pbx.c: Executing [19498138011@phones:1] NoOp(“PJSIP/enigma-00000004”, “19498138011”) in new stack
[2025-06-09 12:02:59.276] VERBOSE[3170134][C-00000003] pbx.c: Executing [19498138011@phones:2] Goto(“PJSIP/enigma-00000004”, “outgoing,19498138011,1”) in new stack
[2025-06-09 12:02:59.277] VERBOSE[3170134][C-00000003] pbx_builtins.c: Goto (outgoing,19498138011,1)
[2025-06-09 12:02:59.277] VERBOSE[3170134][C-00000003] pbx.c: Executing [19498138011@outgoing:1] Set(“PJSIP/enigma-00000004”, “CALLERID(num)=17083957689”) in new stack
[2025-06-09 12:02:59.287] VERBOSE[3170134][C-00000003] pbx.c: Executing [19498138011@outgoing:2] Dial(“PJSIP/enigma-00000004”, “PJSIP/19498138011@provider”) in new stack
[2025-06-09 12:02:59.297] VERBOSE[3170134][C-00000003] app_dial.c: Called PJSIP/19498138011@provider
[2025-06-09 12:02:59.845] VERBOSE[3170134][C-00000003] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2025-06-09 12:02:59.845] VERBOSE[3170134][C-00000003] pbx.c: Executing [19498138011@outgoing:3] Hangup(“PJSIP/enigma-00000004”, “”) in new stack
[2025-06-09 12:02:59.845] VERBOSE[3170134][C-00000003] pbx.c: Spawn extension (outgoing, 19498138011, 3) exited non-zero on ‘PJSIP/enigma-00000004’
[2025-06-09 12:03:40.391] NOTICE[3168237] res_pjsip/pjsip_distributor.c: Request ‘OPTIONS’ from ‘“Censys” sip:censysinspect@censys.io’ failed for ‘162.142.125.244:59313’ (callid: a84b4c76e66710) - No matching endpoint found
[2025-06-09 12:04:39.662] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP request (876 bytes) from UDP:10.0.0.177:42961 —>
INVITE sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—38a1fc691ef06f79;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.177:42961;transport=UDP
To: sip:19498138011@192.168.1.2
From: sip:enigma@192.168.1.2;transport=UDP;tag=0cd4ab78
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.6
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 192

v=0
o=Zoiper 0 556278903 IN IP4 10.0.0.177
s=Zoiper
c=IN IP4 10.0.0.177
t=0 0
m=audio 48534 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

[2025-06-09 12:04:39.665] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP response (505 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—38a1fc691ef06f79
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
From: sip:enigma@192.168.1.2;tag=0cd4ab78
To: sip:19498138011@192.168.1.2;tag=z9hG4bK-524287-1—38a1fc691ef06f79
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749488679/351fce92e8afb11e364e70501e988005”,opaque=“353b7fa710700e4b”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 18.23.1
Content-Length: 0

[2025-06-09 12:04:39.679] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP request (360 bytes) from UDP:10.0.0.177:42961 —>
ACK sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—38a1fc691ef06f79;rport
Max-Forwards: 70
To: sip:19498138011@192.168.1.2;tag=z9hG4bK-524287-1—38a1fc691ef06f79
From: sip:enigma@192.168.1.2;transport=UDP;tag=0cd4ab78
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
CSeq: 1 ACK
Content-Length: 0

[2025-06-09 12:04:39.778] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP request (1181 bytes) from UDP:10.0.0.177:42961 —>
INVITE sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—b9bdc63affa6c131;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.177:42961;transport=UDP
To: sip:19498138011@192.168.1.2
From: sip:enigma@192.168.1.2;transport=UDP;tag=0cd4ab78
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.6
Authorization: Digest username=“enigma”,realm=“asterisk”,nonce=“1749488679/351fce92e8afb11e364e70501e988005”,uri="sip:19498138011@192.168.1.2;transport=UDP",response=“cbef01230be84c17a5dccae7637cdfe1”,cnonce=“8ed1a855cd02377becce6657ce14f6a8”,nc=00000001,qop=auth,algorithm=MD5,opaque=“353b7fa710700e4b”
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 192

v=0
o=Zoiper 0 556278903 IN IP4 10.0.0.177
s=Zoiper
c=IN IP4 10.0.0.177
t=0 0
m=audio 48534 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

[2025-06-09 12:04:39.781] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP response (313 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—b9bdc63affa6c131
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
From: sip:enigma@192.168.1.2;tag=0cd4ab78
To: sip:19498138011@192.168.1.2
CSeq: 2 INVITE
Server: Asterisk PBX 18.23.1
Content-Length: 0

[2025-06-09 12:04:39.785] VERBOSE[3170213][C-00000004] pbx.c: Executing [19498138011@phones:1] NoOp(“PJSIP/enigma-00000006”, “19498138011”) in new stack
[2025-06-09 12:04:39.785] VERBOSE[3170213][C-00000004] pbx.c: Executing [19498138011@phones:2] Goto(“PJSIP/enigma-00000006”, “outgoing,19498138011,1”) in new stack
[2025-06-09 12:04:39.785] VERBOSE[3170213][C-00000004] pbx_builtins.c: Goto (outgoing,19498138011,1)
[2025-06-09 12:04:39.785] VERBOSE[3170213][C-00000004] pbx.c: Executing [19498138011@outgoing:1] Set(“PJSIP/enigma-00000006”, “CALLERID(num)=17083957689”) in new stack
[2025-06-09 12:04:39.786] VERBOSE[3170213][C-00000004] pbx.c: Executing [19498138011@outgoing:2] Dial(“PJSIP/enigma-00000006”, “PJSIP/19498138011@provider”) in new stack
[2025-06-09 12:04:39.788] VERBOSE[3170213][C-00000004] app_dial.c: Called PJSIP/19498138011@provider
[2025-06-09 12:04:40.109] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP request (994 bytes) to UDP:34.226.36.33:5060 —>
INVITE sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 24.13.121.136:5060;rport;branch=z9hG4bKPj141ea8c8-11ec-45c9-97aa-2ae381c9dd47
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com
Contact: sip:asterisk@24.13.121.136:5060
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16725 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length: 261

v=0
o=- 881804186 881804186 IN IP4 24.13.121.136
s=Asterisk
c=IN IP4 24.13.121.136
t=0 0
m=audio 19766 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-06-09 12:04:40.140] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP response (379 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.13.121.136:5060;rport=5060;branch=z9hG4bKPj141ea8c8-11ec-45c9-97aa-2ae381c9dd47;received=24.13.121.136
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16725 INVITE
Content-Length: 0

[2025-06-09 12:04:40.204] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP response (556 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 24.13.121.136:5060;received=24.13.121.136;rport=5060;branch=z9hG4bKPj141ea8c8-11ec-45c9-97aa-2ae381c9dd47
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.caacdd12
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16725 INVITE
Proxy-Authenticate: Digest realm=“sip.flowroute.com”, nonce=“aEcVVGhHFCiELHy9+kOlpFITkBHbppwn”, qop=“auth”
Content-Length: 0

[2025-06-09 12:04:40.205] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP request (485 bytes) to UDP:34.226.36.33:5060 —>
ACK sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 24.13.121.136:5060;rport;branch=z9hG4bKPj141ea8c8-11ec-45c9-97aa-2ae381c9dd47
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.caacdd12
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16725 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0

[2025-06-09 12:04:40.206] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP request (1275 bytes) to UDP:34.226.36.33:5060 —>
INVITE sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 24.13.121.136:5060;rport;branch=z9hG4bKPj21c56d34-5990-454a-b649-55e7cf53983f
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com
Contact: sip:asterisk@24.13.121.136:5060
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16726 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Proxy-Authorization: Digest username=“provider”, realm=“sip.flowroute.com”, nonce=“aEcVVGhHFCiELHy9+kOlpFITkBHbppwn”, uri="sip:19498138011@us-east-va.sip.flowroute.com", response=“912fe0f0a9609d073c56c354b0538bf7”, cnonce=“cc3eb5e1c51541d3b68820d770e17ff7”, qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 261

v=0
o=- 881804186 881804186 IN IP4 24.13.121.136
s=Asterisk
c=IN IP4 24.13.121.136
t=0 0
m=audio 19766 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-06-09 12:04:40.237] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP response (379 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.13.121.136:5060;rport=5060;branch=z9hG4bKPj21c56d34-5990-454a-b649-55e7cf53983f;received=24.13.121.136
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16726 INVITE
Content-Length: 0

[2025-06-09 12:04:40.300] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP response (449 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 403 Bad au - support@flowroute.com
Via: SIP/2.0/UDP 24.13.121.136:5060;received=24.13.121.136;rport=5060;branch=z9hG4bKPj21c56d34-5990-454a-b649-55e7cf53983f
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.5857dd12
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16726 INVITE
Content-Length: 0

[2025-06-09 12:04:40.301] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP request (485 bytes) to UDP:34.226.36.33:5060 —>
ACK sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 24.13.121.136:5060;rport;branch=z9hG4bKPj21c56d34-5990-454a-b649-55e7cf53983f
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=896c6edd-15ca-4b07-b18c-e22b0900d393
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.5857dd12
Call-ID: b5f9aa13-b645-40c8-b071-43e838648619
CSeq: 16726 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0

[2025-06-09 12:04:40.302] VERBOSE[3170213][C-00000004] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2025-06-09 12:04:40.302] VERBOSE[3170213][C-00000004] pbx.c: Executing [19498138011@outgoing:3] Hangup(“PJSIP/enigma-00000006”, “”) in new stack
[2025-06-09 12:04:40.302] VERBOSE[3170213][C-00000004] pbx.c: Spawn extension (outgoing, 19498138011, 3) exited non-zero on ‘PJSIP/enigma-00000006’
[2025-06-09 12:04:40.303] VERBOSE[3168237] res_pjsip_logger.c: <— Transmitting SIP response (381 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—b9bdc63affa6c131
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
From: sip:enigma@192.168.1.2;tag=0cd4ab78
To: sip:19498138011@192.168.1.2;tag=92ed181d-419e-4081-83c0-f1424e73f992
CSeq: 2 INVITE
Server: Asterisk PBX 18.23.1
Reason: Q.850;cause=21
Content-Length: 0

[2025-06-09 12:04:40.318] VERBOSE[3168236] res_pjsip_logger.c: <— Received SIP request (361 bytes) from UDP:10.0.0.177:42961 —>
ACK sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—b9bdc63affa6c131;rport
Max-Forwards: 70
To: sip:19498138011@192.168.1.2;tag=92ed181d-419e-4081-83c0-f1424e73f992
From: sip:enigma@192.168.1.2;transport=UDP;tag=0cd4ab78
Call-ID: pJ6HNtVnbeZKBMsEFNPRWw..
CSeq: 2 ACK
Content-Length: 0

----- Note I disable the firewall on both the asterisk server and router firewall, and still no communication both inbound or outbound just do not work. ------

2

Unlikely

outbound_auth is unlikely to be used.

Security fix only status and only about 4 months left of that.

3

Just updated to asterisk 22.4.1 and I still get the same problem. NO inbound or outbound calls. What if any would I have to change in new config. I use sip_to_pjsip.py script to convert.

The output was the above pjsip.conf.

my extensions.conf

[globals]

[phones]
exten => 200,1,NoOp()
same => n,Dial(PJSIP/hulk,20)
same => n,VoiceMail(200)
same => n,Hangup()

exten => *200,1,NoOp()
same => n,VoiceMailMain(200)
same => n,Hangup()

exten => 100,1,NoOp()
same => n,Dial(PJSIP/enigma,20)
same => n,VoiceMail(100)
same => n,HangUp()

exten => *100,1,NoOp()
same => n,VoiceMailMain(100)
same => n,Hangup()

exten => 500,1,NoOp()
same => n,Answer()
same => n,Playback(demo-congrats)
same => n,Wait(2)
same => n,Congestion()
same => n,HangUp()

exten => _X.,1,NoOp(${EXTEN})
same => n,Goto(outgoing,${EXTEN},1)

[outgoing]
;exten => _XXXXXXXXXXX,1,Log(Notice,External call from ${CALLERID(num)} to ${EXTEN})

exten => _X.,1,Set(CALLERID(num)=17083957689
same => n,Dial(PJSIP/${EXTEN}@provider)
;same => n,Congestion()
same => n,Hangup()

[provider]
exten => _X.,1,Goto(phones,200,1)

This is the asterisk pjsip set logger on output
yet I don’t know the cause. I think this is the problem the part
that says.

<— Received SIP response (446 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 403 Bad au - support@flowroute.com
Via: SIP/2.0/UDP 192.168.1.2:5060;received=24.13.121.136;rport=5060;branch=z9hG4bKPjbf6acc22-664e-4e67-ad55-7d365e8160d9
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.95e6e09e
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4780 INVITE
Content-Length: 0

-----------At The of the communication I get a FORBIDDEN HERE GO THE PACKETS----------------------

<— Received SIP request (876 bytes) from UDP:10.0.0.177:42961 —>
INVITE sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—c169fe56400d4d77;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.177:42961;transport=UDP
To: sip:19498138011@192.168.1.2
From: sip:enigma@192.168.1.2;transport=UDP;tag=8055da15
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.6
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 192

v=0
o=Zoiper 0 581991053 IN IP4 10.0.0.177
s=Zoiper
c=IN IP4 10.0.0.177
t=0 0
m=audio 37009 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<— Transmitting SIP response (504 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—c169fe56400d4d77
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
From: sip:enigma@192.168.1.2;tag=8055da15
To: sip:19498138011@192.168.1.2;tag=z9hG4bK-524287-1—c169fe56400d4d77
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749517167/f5f7d44cd3816e2d32dec599d021d944”,opaque=“682213043ba6c31b”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Received SIP request (360 bytes) from UDP:10.0.0.177:42961 —>
ACK sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—c169fe56400d4d77;rport
Max-Forwards: 70
To: sip:19498138011@192.168.1.2;tag=z9hG4bK-524287-1—c169fe56400d4d77
From: sip:enigma@192.168.1.2;transport=UDP;tag=8055da15
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1181 bytes) from UDP:10.0.0.177:42961 —>
INVITE sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—b976e8bd2c3de259;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.177:42961;transport=UDP
To: sip:19498138011@192.168.1.2
From: sip:enigma@192.168.1.2;transport=UDP;tag=8055da15
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.6
Authorization: Digest username=“enigma”,realm=“asterisk”,nonce=“1749517167/f5f7d44cd3816e2d32dec599d021d944”,uri="sip:19498138011@192.168.1.2;transport=UDP",response=“b785b8e855dde30947e5f42114d7d139”,cnonce=“aa3c40081d078d08315ab82223983b93”,nc=00000001,qop=auth,algorithm=MD5,opaque=“682213043ba6c31b”
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 192

v=0
o=Zoiper 0 581991053 IN IP4 10.0.0.177
s=Zoiper
c=IN IP4 10.0.0.177
t=0 0
m=audio 37009 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<— Transmitting SIP response (312 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—b976e8bd2c3de259
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
From: sip:enigma@192.168.1.2;tag=8055da15
To: sip:19498138011@192.168.1.2
CSeq: 2 INVITE
Server: Asterisk PBX 22.4.1
Content-Length: 0

-- Executing [19498138011@phones:1] NoOp("PJSIP/enigma-00000007", "19498138011") in new stack
-- Executing [19498138011@phones:2] Goto("PJSIP/enigma-00000007", "outgoing,19498138011,1") in new stack
-- Goto (outgoing,19498138011,1)
-- Executing [19498138011@outgoing:1] Set("PJSIP/enigma-00000007", "CALLERID(num)=17083957689") in new stack
-- Executing [19498138011@outgoing:2] Dial("PJSIP/enigma-00000007", "PJSIP/19498138011@provider") in new stack
-- Called PJSIP/19498138011@provider

<— Transmitting SIP request (986 bytes) to UDP:34.226.36.33:5060 —>
INVITE sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKPj408b930d-ff8a-46f2-b6f2-c2d4722e1430
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com
Contact: sip:asterisk@192.168.1.2:5060
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4779 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.4.1
Content-Type: application/sdp
Content-Length: 259

v=0
o=- 1887634874 1887634874 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 19550 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP response (376 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bKPj408b930d-ff8a-46f2-b6f2-c2d4722e1430;received=24.13.121.136
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4779 INVITE
Content-Length: 0

<— Received SIP response (553 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5060;received=24.13.121.136;rport=5060;branch=z9hG4bKPj408b930d-ff8a-46f2-b6f2-c2d4722e1430
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.9e2ae09e
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4779 INVITE
Proxy-Authenticate: Digest realm=“sip.flowroute.com”, nonce=“aEeEtGhHg4hHtgSH2xPfwil4K0QHYnLg”, qop=“auth”
Content-Length: 0

<— Transmitting SIP request (481 bytes) to UDP:34.226.36.33:5060 —>
ACK sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKPj408b930d-ff8a-46f2-b6f2-c2d4722e1430
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.9e2ae09e
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4779 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.4.1
Content-Length: 0

<— Transmitting SIP request (1267 bytes) to UDP:34.226.36.33:5060 —>
INVITE sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKPjbf6acc22-664e-4e67-ad55-7d365e8160d9
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com
Contact: sip:asterisk@192.168.1.2:5060
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4780 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.4.1
Proxy-Authorization: Digest username=“provider”, realm=“sip.flowroute.com”, nonce=“aEeEtGhHg4hHtgSH2xPfwil4K0QHYnLg”, uri="sip:19498138011@us-east-va.sip.flowroute.com", response=“7e78dcfa1e0d81b6f0dab961c241e06c”, cnonce=“7c98546f5f04469c8decce9b314b2364”, qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 259

v=0
o=- 1887634874 1887634874 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 19550 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP response (376 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bKPjbf6acc22-664e-4e67-ad55-7d365e8160d9;received=24.13.121.136
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4780 INVITE
Content-Length: 0

<— Received SIP response (446 bytes) from UDP:34.226.36.33:5060 —>
SIP/2.0 403 Bad au - support@flowroute.com
Via: SIP/2.0/UDP 192.168.1.2:5060;received=24.13.121.136;rport=5060;branch=z9hG4bKPjbf6acc22-664e-4e67-ad55-7d365e8160d9
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.95e6e09e
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4780 INVITE
Content-Length: 0

<— Transmitting SIP request (481 bytes) to UDP:34.226.36.33:5060 —>
ACK sip:19498138011@us-east-va.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKPjbf6acc22-664e-4e67-ad55-7d365e8160d9
From: sip:17083957689@us-east-va.sip.flowroute.com;tag=3847e25b-6fa8-467e-853d-c854fa0698b7
To: sip:19498138011@us-east-va.sip.flowroute.com;tag=bf8638324618dc61059d4c604476fea1.95e6e09e
Call-ID: 1e03ee95-ff5b-4f4b-834d-a432e5d38bae
CSeq: 4780 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.4.1
Content-Length: 0

== Everyone is busy/congested at this time (1:0/0/1)
– Executing [19498138011@outgoing:3] Hangup(“PJSIP/enigma-00000007”, “”) in new stack
== Spawn extension (outgoing, 19498138011, 3) exited non-zero on ‘PJSIP/enigma-00000007’
<— Transmitting SIP response (380 bytes) to UDP:10.0.0.177:42961 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.177:42961;rport=42961;received=10.0.0.177;branch=z9hG4bK-524287-1—b976e8bd2c3de259
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
From: sip:enigma@192.168.1.2;tag=8055da15
To: sip:19498138011@192.168.1.2;tag=8391bae7-660f-4a51-bae3-8fc91fff3104
CSeq: 2 INVITE
Server: Asterisk PBX 22.4.1
Reason: Q.850;cause=21
Content-Length: 0

<— Received SIP request (361 bytes) from UDP:10.0.0.177:42961 —>
ACK sip:19498138011@192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.177:42961;branch=z9hG4bK-524287-1—b976e8bd2c3de259;rport
Max-Forwards: 70
To: sip:19498138011@192.168.1.2;tag=8391bae7-660f-4a51-bae3-8fc91fff3104
From: sip:enigma@192.168.1.2;transport=UDP;tag=8055da15
Call-ID: LC_IWoe5nLMnzGSa9tmRxg..
CSeq: 2 ACK
Content-Length: 0

4

You need to provide the correct authentication data. Given that username looks wrong to me, I assume au mean authentication user. In any case you would need to contact the support address given, to be sure.

I’d expect it to be your account or phone number.

5

Where at do I change the username to account or phone number because I thought I was doing that with Set(callerid).

6

I talked to my SIP support and they said it may be a NAT problem.
My asterisk server is wired connected to a LAN of 192.168.1.0/24
and my sip endpoints are connected to wireless LAN of 10.0.0.0/24.
Now I can register the endpoints and make calls between them.
However, any communication directed outside the network or
incoming calls just do not work.

7

Where you currently set it to “provider”!

8

Thanks a lot. Changing provider to my sip username worked. However, two more problems come up when somebody answer. Here is the asterisk output.

Spawn extension (outgoing, 19498138011, 2) exited non-zero on ‘PJSIP/hulk-0000000c’
– Executing [17086326111@phones:1] NoOp(“PJSIP/hulk-0000000e”, “17086326111”) in new stack
– Executing [17086326111@phones:2] Goto(“PJSIP/hulk-0000000e”, “outgoing,17086326111,1”) in new stack
– Goto (outgoing,17086326111,1)
– Executing [17086326111@outgoing:1] Set(“PJSIP/hulk-0000000e”, “CALLERID(num)=17083957689”) in new stack
– Executing [17086326111@outgoing:2] Dial(“PJSIP/hulk-0000000e”, “PJSIP/17086326111@provider”) in new stack
– Called PJSIP/17086326111@provider
> 0x7f480b9f30 – Strict RTP learning after remote address set to: 216.82.225.140:48488
[Jun 10 15:21:44] WARNING[47517]: channel.c:5703 set_format: Unable to find a codec translation path: (ulaw) → (alaw)
[Jun 10 15:21:44] WARNING[47517]: channel.c:5703 set_format: Unable to find a codec translation path: (alaw) → (ulaw)
– PJSIP/provider-0000000f is making progress passing it to PJSIP/hulk-0000000e
> 0x7f480bf0d0 – Strict RTP learning after remote address set to: 10.0.0.3:48223
– PJSIP/provider-0000000f is making progress passing it to PJSIP/hulk-0000000e
– PJSIP/provider-0000000f answered PJSIP/hulk-0000000e
[Jun 10 15:21:54] WARNING[48235][C-00000008]: channel.c:6693 ast_channel_make_compatible_helper: No path to translate from PJSIP/provider-0000000f to PJSIP/hulk-0000000e
[Jun 10 15:21:54] WARNING[48235][C-00000008]: app_dial.c:3383 dial_exec_full: Had to drop call because I couldn’t make PJSIP/hulk-0000000e compatible with PJSIP/provider-0000000f
== Spawn extension (outgoing, 17086326111, 2) exited non-zero on ‘PJSIP/hulk-0000000e’

9

You are missing a codec module, which suggests you are trying to load a minimal set of modules, as the one you are missing is not one that people would normally omit, so I can’t remember its name.

It’s better to use autoload=yes, and then try using noload, selectively, once it works. One side is using the North American version of G.711, and the other is using the European, although most people just load everything. Unless you are breaking out in multiple continents, you would be better just to enable the appropriate one for your location µ-law is North America, and A-law is Europe.

10

Thank a lot David551. It worked. However I can not hear conversation on either phone when call is accepted. Just for testing purposes I am allow all outbound traffic on my firewall. Here is the output of asterisk.

Executing [17086326111@phones:1] NoOp(“PJSIP/hulk-00000004”, “17086326111”) in new stack
– Executing [17086326111@phones:2] Goto(“PJSIP/hulk-00000004”, “outgoing,17086326111,1”) in new stack
– Goto (outgoing,17086326111,1)
– Executing [17086326111@outgoing:1] Set(“PJSIP/hulk-00000004”, “CALLERID(num)=17083957689”) in new stack
– Executing [17086326111@outgoing:2] Dial(“PJSIP/hulk-00000004”, “PJSIP/17086326111@provider”) in new stack
– Called PJSIP/17086326111@provider
> 0x7f3c22f980 – Strict RTP learning after remote address set to: 67.231.3.94:53352
– PJSIP/provider-00000005 is making progress passing it to PJSIP/hulk-00000004
> 0x7f3c2285b0 – Strict RTP learning after remote address set to: 10.0.0.3:44386
– PJSIP/provider-00000005 is making progress passing it to PJSIP/hulk-00000004
– PJSIP/provider-00000005 answered PJSIP/hulk-00000004
– Channel PJSIP/provider-00000005 joined ‘simple_bridge’ basic-bridge <32198bf4-ece3-4a9f-ac3e-27ae7956c474>
– Channel PJSIP/hulk-00000004 joined ‘simple_bridge’ basic-bridge <32198bf4-ece3-4a9f-ac3e-27ae7956c474>
– Channel PJSIP/hulk-00000004 left ‘simple_bridge’ basic-bridge <32198bf4-ece3-4a9f-ac3e-27ae7956c474>
– Channel PJSIP/provider-00000005 left ‘simple_bridge’ basic-bridge <32198bf4-ece3-4a9f-ac3e-27ae7956c474>
== Spawn extension (outgoing, 17086326111, 2) exited non-zero on ‘PJSIP/hulk-00000004’

11

Try disabling direct media, at least on the provider endpoint. if that doesn’t work, turn on RTP debugging.

12

Already had direct media disable. However when I turn on rtp set debug on. I only get one packet, it looks like.

Called PJSIP/17086326111@provider
> 0x7f404abfd0 – Strict RTP learning after remote address set to: 216.82.225.138:52146
– PJSIP/provider-00000003 is making progress passing it to PJSIP/hulk-00000002
> 0x7f4030ae00 – Strict RTP learning after remote address set to: 10.0.0.3:32787

13

That doesn’t show any.

Why do you have this?

14

This is my current pjsip.conf

[general]
allowoverlap = no

[provider]
insecure = invite

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
–;

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0

local_net = 192.168.1.0/24
local_net = 10.0.0.0/24

[reg_us-east-va.sip.flowroute.com]
type = registration
retry_interval = 20
max_retries = 10
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_us-east-va.sip.flowroute.com
client_uri = sip:xxxxxxxxxx@us-east-va.sip.flowroute.com
server_uri = sip:us-east-va.sip.flowroute.com

[auth_reg_us-east-va.sip.flowroute.com]
type = auth
password = xxxxxxxxxxxx
username = xxxxxxxxx

[hulk]
type = aor
max_contacts = 1

[hulk]
type = auth
username = hulk
auth_type = userpass
password = xxxxxxxxxxxx

[hulk]
type = endpoint
context = phones
allow = alaw,ulaw
auth = hulk
transport = transport-udp
outbound_auth = hulk
auth = hulk
aors = hulk

[enigma]
type = aor
max_contacts = 1

[enigma]
type = auth
username = enigma
auth_type = userpass
password = xxxxxxxxxx

[enigma]
type = endpoint
context = phones
allow = alaw,ulaw
transport = transport-udp
auth = enigma
outbound_auth = enigma
aors = enigma

[provider]
type = aor
contact = sip:xxxxxxxx@us-east-va.sip.flowroute.com:5060

[provider]
type = identify
endpoint = provider
match = us-east-va.sip.flowroute.com

[provider]
type = auth
username = xxxxxxxxx
password = xxxxxxxxx

[provider]
type = endpoint
transport=transport-udp
context = outgoing
disallow = all
allow = ulaw,alaw
transport = transport-udp
;rtp_symmetric = yes
;force_rport = yes
;rewrite_contact = yes
from_domain = us-east-va.sip.flowroute.com
auth = provider
outbound_auth = provider
direct_media = no
;contact_user = 200
aors = provider

Here is a tshark capture of my call

739 30.334852904 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18475, Time=4211619718
740 30.372691906 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18476, Time=4211619878
741 30.372692239 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18477, Time=4211620038
742 30.384855612 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18478, Time=4211620198
743 30.406341037 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18479, Time=4211620358
744 30.424239755 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18480, Time=4211620518
745 30.455556155 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18481, Time=4211620678
746 30.477496819 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18482, Time=4211620838
747 30.494308819 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18483, Time=4211620998
748 30.528371356 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18484, Time=4211621158
749 30.528371726 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18485, Time=4211621318
750 30.556141920 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18486, Time=4211621478
751 30.578557452 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18487, Time=4211621638
752 30.612807506 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18488, Time=4211621798
753 30.615476624 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18489, Time=4211621958
754 30.634632725 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18490, Time=4211622118
755 30.666361624 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18491, Time=4211622278
756 30.666361976 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18492, Time=4211622438
757 30.694272335 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18493, Time=4211622598
758 30.730561640 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18494, Time=4211622758
759 30.736764113 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18495, Time=4211622918
760 30.755202125 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18496, Time=4211623078
761 30.782616487 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18497, Time=4211623238
762 30.782616820 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18498, Time=4211623398
763 30.804873167 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18499, Time=4211623558
764 30.834301631 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18500, Time=4211623718
765 30.844271995 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18501, Time=4211623878
766 30.885765536 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18502, Time=4211624038
767 30.894299869 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18503, Time=4211624198
768 30.919682110 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18504, Time=4211624358
769 30.919682444 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18505, Time=4211624518
770 30.944928148 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18506, Time=4211624678
771 30.996096758 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18507, Time=4211624838
772 30.996097092 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18508, Time=4211624998
773 31.038982034 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18509, Time=4211625158
774 31.038982367 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18510, Time=4211625318
775 31.057120585 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18511, Time=4211625478
776 31.097860425 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18512, Time=4211625638
777 31.097860814 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18513, Time=4211625798
778 31.117428710 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18514, Time=4211625958
779 31.149049943 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18515, Time=4211626118
780 31.155204693 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18516, Time=4211626278
781 31.199433426 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18517, Time=4211626438
782 31.199433741 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18518, Time=4211626598
783 31.216885646 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18519, Time=4211626758
784 31.216885961 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18520, Time=4211626918
785 31.244870987 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18521, Time=4211627078
786 31.264868289 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18522, Time=4211627238
787 31.302422088 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18523, Time=4211627398
788 31.315987195 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18524, Time=4211627558
789 31.353712790 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18525, Time=4211627718
790 31.356101335 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18526, Time=4211627878
791 31.356101631 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18527, Time=4211628038
792 31.404385791 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18528, Time=4211628198
793 31.404920103 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18529, Time=4211628358
794 31.424851553 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18530, Time=4211628518
795 31.455880214 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18531, Time=4211628678
796 31.474796632 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18532, Time=4211628838
797 31.495125395 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18533, Time=4211628998
798 31.495125765 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18534, Time=4211629158
799 31.524987709 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18535, Time=4211629318
800 31.558472600 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18536, Time=4211629478
801 31.565233237 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18537, Time=4211629638
802 31.608660529 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18538, Time=4211629798
803 31.614914576 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18539, Time=4211629958
804 31.660592858 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18540, Time=4211630118
805 31.660593191 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18541, Time=4211630278
806 31.664873580 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18542, Time=4211630438
807 31.684348569 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18543, Time=4211630598
808 31.711615561 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18544, Time=4211630758
809 31.738921182 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18545, Time=4211630918
810 31.754883557 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18546, Time=4211631078
811 31.775047673 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18547, Time=4211631238
812 31.775048006 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18548, Time=4211631398
813 31.814828295 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18549, Time=4211631558
814 31.828792197 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18550, Time=4211631718
815 31.866094053 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18551, Time=4211631878
816 31.874867959 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18552, Time=4211632038
817 31.916276019 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18553, Time=4211632198
818 31.916276371 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18554, Time=4211632358
819 31.916276538 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18555, Time=4211632518
820 31.968791364 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18556, Time=4211632678
821 31.971499223 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18557, Time=4211632838
822 31.986543287 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18558, Time=4211632998
823 32.014916237 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18559, Time=4211633158
824 32.035879756 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18560, Time=4211633318
825 32.071164769 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18561, Time=4211633478
826 32.074219274 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18562, Time=4211633638
827 32.121009200 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18563, Time=4211633798
828 32.121009552 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18564, Time=4211633958
829 32.139221694 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18565, Time=4211634118
830 32.173984468 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18566, Time=4211634278
831 32.178335283 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18567, Time=4211634438
832 32.195287894 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18568, Time=4211634598
833 32.215087345 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18569, Time=4211634758
834 32.215087678 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18570, Time=4211634918
835 32.276423077 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18571, Time=4211635078
836 32.276423577 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18572, Time=4211635238
837 32.327115226 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x4A8D7305, Seq=18573, Time=4211635398
838 32.327115578 10.0.0.3 ? 192.168.1.2 RTP 214 PT=ITU-T G.711 PCMA,

15

I figured it out. I had to set up some inbound firewall rules for rtp ports to asterisk server.
Now on to the Inbound calls. Do you see anything that may stop inbound calls?

16

You can’t have local_net without also having public media and signalling addresses. You need those as you have NAT, although the provider might work round that, but if they do, you will have to be the first to send media.

17

I just enable these two setting, yet it still does not work
external_media_address=x.x.x.x.x.x
external_signaling_address=x.x.x.x.x.x
Is this right for public media.

Here is a dump with logging on, so the packets are making it to the server.

<— Received SIP request (1137 bytes) from UDP:34.226.36.35:5060 —>
INVITE sip:17083957689@192.168.1.2:5060 SIP/2.0
Record-Route: sip:34.226.36.35;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
Record-Route: sip:34.211.73.216;lr
To: sip:+17083957689@fl.gg
Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK4cdf.6810cb9112e2087c63e836975616280d.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.2
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 INVITE
Contact: sip:+17086326111@74.120.93.30:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 219
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: “[V]JEFFERSON TI” sip:+17086326111;verstat=TN-Validation-Passed@fl.gg
P-Attestation-Indicator: A

v=0
o=- 591018 855974 IN IP4 74.120.93.29
s=-
c=IN IP4 74.120.93.29
t=0 0
m=audio 49186 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<— Transmitting SIP response (880 bytes) to UDP:34.226.36.35:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK4cdf.6810cb9112e2087c63e836975616280d.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.2
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Record-Route: sip:34.226.36.35;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 50373426_127920774@74.120.93.30
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.6810cb9112e2087c63e836975616280d.0
CSeq: 348007 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749654602/3cb9bc160f5e7d64fa8885d1739eea98”,opaque=“3bef21441046cf1f”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Received SIP request (381 bytes) from UDP:34.226.36.35:5060 —>
ACK sip:17083957689@192.168.1.2:5060 SIP/2.0
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.6810cb9112e2087c63e836975616280d.0
Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK4cdf.6810cb9112e2087c63e836975616280d.0
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 ACK
Content-Length: 0

<— Received SIP request (1137 bytes) from UDP:34.226.36.34:5060 —>
INVITE sip:17083957689@192.168.1.2:5060 SIP/2.0
Record-Route: sip:34.226.36.34;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
Record-Route: sip:34.211.73.216;lr
To: sip:+17083957689@fl.gg
Via: SIP/2.0/UDP 34.226.36.34:5060;branch=z9hG4bK4cdf.faef23dcbc66b6728d91e4299b4ba3e2.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.4
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 INVITE
Contact: sip:+17086326111@74.120.93.30:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 219
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: “[V]JEFFERSON TI” sip:+17086326111;verstat=TN-Validation-Passed@fl.gg
P-Attestation-Indicator: A

v=0
o=- 591018 855974 IN IP4 74.120.93.29
s=-
c=IN IP4 74.120.93.29
t=0 0
m=audio 49186 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<— Received SIP request (1139 bytes) from UDP:34.226.36.34:5060 —>
INVITE sip:17083957689@24.13.121.136:5060 SIP/2.0
Record-Route: sip:34.226.36.34;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
Record-Route: sip:34.211.73.216;lr
To: sip:+17083957689@fl.gg
Via: SIP/2.0/UDP 34.226.36.34:5060;branch=z9hG4bK4cdf.158c1d5c7160e95b55c8031514044ffa.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.3
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 INVITE
Contact: sip:+17086326111@74.120.93.30:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 219
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: “[V]JEFFERSON TI” sip:+17086326111;verstat=TN-Validation-Passed@fl.gg
P-Attestation-Indicator: A

v=0
o=- 591018 855974 IN IP4 74.120.93.29
s=-
c=IN IP4 74.120.93.29
t=0 0
m=audio 49186 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<— Transmitting SIP response (880 bytes) to UDP:34.226.36.34:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 34.226.36.34:5060;rport=5060;received=34.226.36.34;branch=z9hG4bK4cdf.faef23dcbc66b6728d91e4299b4ba3e2.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.4
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Record-Route: sip:34.226.36.34;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 50373426_127920774@74.120.93.30
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.faef23dcbc66b6728d91e4299b4ba3e2.0
CSeq: 348007 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749654602/2b597a0886c1434bbd2bccd2c776561c”,opaque=“13c85c4a49cf42d6”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Transmitting SIP response (880 bytes) to UDP:34.226.36.34:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 34.226.36.34:5060;rport=5060;received=34.226.36.34;branch=z9hG4bK4cdf.158c1d5c7160e95b55c8031514044ffa.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.3
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Record-Route: sip:34.226.36.34;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 50373426_127920774@74.120.93.30
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.158c1d5c7160e95b55c8031514044ffa.0
CSeq: 348007 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749654602/2b597a0886c1434bbd2bccd2c776561c”,opaque=“715d5b0a6e19c271”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Received SIP request (383 bytes) from UDP:34.226.36.34:5060 —>
ACK sip:17083957689@24.13.121.136:5060 SIP/2.0
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.158c1d5c7160e95b55c8031514044ffa.0
Via: SIP/2.0/UDP 34.226.36.34:5060;branch=z9hG4bK4cdf.158c1d5c7160e95b55c8031514044ffa.0
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 ACK
Content-Length: 0

<— Received SIP request (381 bytes) from UDP:34.226.36.34:5060 —>
ACK sip:17083957689@192.168.1.2:5060 SIP/2.0
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.faef23dcbc66b6728d91e4299b4ba3e2.0
Via: SIP/2.0/UDP 34.226.36.34:5060;branch=z9hG4bK4cdf.faef23dcbc66b6728d91e4299b4ba3e2.0
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 ACK
Content-Length: 0

<— Received SIP request (1137 bytes) from UDP:34.226.36.33:5060 —>
INVITE sip:17083957689@192.168.1.2:5060 SIP/2.0
Record-Route: sip:34.226.36.33;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
Record-Route: sip:34.211.73.216;lr
To: sip:+17083957689@fl.gg
Via: SIP/2.0/UDP 34.226.36.33:5060;branch=z9hG4bK4cdf.6addb335f88535824ef38f20e35bb4d8.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.5
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 INVITE
Contact: sip:+17086326111@74.120.93.30:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 219
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: “[V]JEFFERSON TI” sip:+17086326111;verstat=TN-Validation-Passed@fl.gg
P-Attestation-Indicator: A

v=0
o=- 591018 855974 IN IP4 74.120.93.29
s=-
c=IN IP4 74.120.93.29
t=0 0
m=audio 49186 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<— Transmitting SIP response (880 bytes) to UDP:34.226.36.33:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 34.226.36.33:5060;rport=5060;received=34.226.36.33;branch=z9hG4bK4cdf.6addb335f88535824ef38f20e35bb4d8.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.5
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Record-Route: sip:34.226.36.33;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 50373426_127920774@74.120.93.30
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.6addb335f88535824ef38f20e35bb4d8.0
CSeq: 348007 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749654602/0f324f70e2052ebb75be927989d6e299”,opaque=“5c44c3745c3270f6”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Received SIP request (381 bytes) from UDP:34.226.36.33:5060 —>
ACK sip:17083957689@192.168.1.2:5060 SIP/2.0
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.6addb335f88535824ef38f20e35bb4d8.0
Via: SIP/2.0/UDP 34.226.36.33:5060;branch=z9hG4bK4cdf.6addb335f88535824ef38f20e35bb4d8.0
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 ACK
Content-Length: 0

<— Received SIP request (1137 bytes) from UDP:34.226.36.32:5060 —>
INVITE sip:17083957689@192.168.1.2:5060 SIP/2.0
Record-Route: sip:34.226.36.32;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
Record-Route: sip:34.211.73.216;lr
To: sip:+17083957689@fl.gg
Via: SIP/2.0/UDP 34.226.36.32:5060;branch=z9hG4bK4cdf.cf4beaafc03f41e60a35615db3b72d31.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.6
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 INVITE
Contact: sip:+17086326111@74.120.93.30:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 219
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: “[V]JEFFERSON TI” sip:+17086326111;verstat=TN-Validation-Passed@fl.gg
P-Attestation-Indicator: A

v=0
o=- 591018 855974 IN IP4 74.120.93.29
s=-
c=IN IP4 74.120.93.29
t=0 0
m=audio 49186 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<— Transmitting SIP response (880 bytes) to UDP:34.226.36.32:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 34.226.36.32:5060;rport=5060;received=34.226.36.32;branch=z9hG4bK4cdf.cf4beaafc03f41e60a35615db3b72d31.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4cdf.b40edb4addd8ed37d4a2dbcb0456c827.6
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK4cdf.c7a3169c15db82dd44b0d1ac725fd542.0
Via: SIP/2.0/UDP 74.120.93.30:5060;branch=z9hG4bK00B50c8d5dedf0444f5
Record-Route: sip:34.226.36.32;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 50373426_127920774@74.120.93.30
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.cf4beaafc03f41e60a35615db3b72d31.0
CSeq: 348007 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1749654602/91dbda641dfe32825f2bf57d21fe0ee3”,opaque=“59b3ad78171551db”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0

<— Received SIP request (381 bytes) from UDP:34.226.36.32:5060 —>
ACK sip:17083957689@192.168.1.2:5060 SIP/2.0
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0060de88
Max-Forwards: 66
To: sip:+17083957689@fl.gg;tag=z9hG4bK4cdf.cf4beaafc03f41e60a35615db3b72d31.0
Via: SIP/2.0/UDP 34.226.36.32:5060;branch=z9hG4bK4cdf.cf4beaafc03f41e60a35615db3b72d31.0
Call-ID: 50373426_127920774@74.120.93.30
CSeq: 348007 ACK
Content-Length: 0

18

You don’t have a type=identify section and the From user seems to be the caller ID, not the identity of the provider.

19

Is not this the type = identify section, and how do I change the from user to identity of provider?

[provider]
type = identify
endpoint = provider
match = us-east-va.sip.flowroute.com
[/quote]

20

I must have mistyped it, when searching for it.

You are trying to authenticate the provider, but I don’t know of any provider that is prepared to authenticate itself! Whilst, in most cases, this doubling is harmless for incoming authentication, it is fatal if the provider only accepts authentication.