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. ------