Thanks a lot David. Incoming calls is working now after deleting the auth = provider. However, I kept the outbound_auth = provider. How do I find more information on the meaning of config options for pjsip.conf like outbound_auth and auth, and also How the whole sip process go, so I can be proficient like you. I appreciate any additional information you can give.
Thanks David. Appreciate all the help.
Sorry to bother you, but I just notice that I can’t hear any sound when I receive incoming calls.
I disable firewall and I still can’t heard any communication. What do you think is the issue. I have communication with outbound calls and from the local network with endpoint to endpoint.
in your rtp.conf file make sure you have 10000-35000 udp
Also don’t turn off firewall instead open your RTP ports.
Thanks for responding, however, I tried that and it still does not work. No voice communication.
If you’d like I can take a look using AnyDesk maybe I can spot something that you can’t
I am on a raspberry pi CLI only. Out going calls works find just incoming with no audio
Still can not hear audio with incoming calls. I get audio with outgoing calls, but not with incoming even when I disable my firewall. However in the pjsip logger message I get an error, no sound. Here is the pjsip log.
<— Received SIP request (1144 bytes) from UDP:34.226.36.35:5060 —>
INVITE sip:17083957689@24.13.121.136:5060 SIP/2.0
Record-Route: sip:34.226.36.35;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
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=z9hG4bK07a8.0882cb3f2eb229db94a054a30cb45f8d.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK07a8.6ed3cd5056cda5a1293613975fdbaf1e.2
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK07a8.3e0eaa7f19d0bbd94635821a9a7931e5.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB52dd2722d60bd32f
Call-ID: 221000176_79612347@206.147.88.72
CSeq: 3003 INVITE
Contact: sip:+17086326111@206.147.88.72:5060
Session-Expires: 1800
Min-SE: 90
Content-Length: 223
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=- 735380 357395 IN IP4 206.147.88.69
s=-
c=IN IP4 206.147.88.69
t=0 0
m=audio 31484 RTP/AVP 0 8 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 (677 bytes) to UDP:34.226.36.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK07a8.0882cb3f2eb229db94a054a30cb45f8d.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK07a8.6ed3cd5056cda5a1293613975fdbaf1e.2
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK07a8.3e0eaa7f19d0bbd94635821a9a7931e5.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB52dd2722d60bd32f
Record-Route: sip:34.226.36.35;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 221000176_79612347@206.147.88.72
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
To: sip:+17083957689@fl.gg
CSeq: 3003 INVITE
Server: Asterisk PBX 22.4.1
Content-Length: 0
-- Executing [17083957689@incoming:1] Goto("PJSIP/provider-00000018", "phones,200,1") in new stack
-- Goto (phones,200,1)
-- Executing [200@phones:1] NoOp("PJSIP/provider-00000018", "") in new stack
-- Executing [200@phones:2] Dial("PJSIP/provider-00000018", "PJSIP/hulk,20") in new stack
[Jun 16 17:53:45] ERROR[5842]: res_pjsip.c:993 ast_sip_create_dialog_uac: Endpoint ‘hulk’: Could not create dialog to invalid URI ‘hulk’. Is endpoint registered and reachable?
[Jun 16 17:53:45] ERROR[5842]: chan_pjsip.c:2702 request: Failed to create outgoing session to endpoint ‘hulk’
[Jun 16 17:53:45] NOTICE[6358][C-00000017]: app_dial.c:2711 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [200@phones:3] VoiceMail(“PJSIP/provider-00000018”, “200”) in new stack
> 0x7f6c028750 – Strict RTP learning after remote address set to: 206.147.88.69:31484
<— Transmitting SIP response (1256 bytes) to UDP:34.226.36.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK07a8.0882cb3f2eb229db94a054a30cb45f8d.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK07a8.6ed3cd5056cda5a1293613975fdbaf1e.2
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK07a8.3e0eaa7f19d0bbd94635821a9a7931e5.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB52dd2722d60bd32f
Record-Route: sip:34.226.36.35;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 221000176_79612347@206.147.88.72
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
To: sip:+17083957689@fl.gg;tag=7af9c92d-6948-4b21-a291-d9299cec4d1e
CSeq: 3003 INVITE
Server: Asterisk PBX 22.4.1
Contact: sip:24.13.121.136:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 255
v=0
o=- 735380 357397 IN IP4 24.13.121.136
s=Asterisk
c=IN IP4 24.13.121.136
t=0 0
m=audio 18834 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
<— Received SIP request (597 bytes) from UDP:34.226.36.35:5060 —>
ACK sip:24.13.121.136:5060 SIP/2.0
Record-Route: sip:34.226.36.35;lr
Max-Forwards: 67
Record-Route: sip:34.211.73.216;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
To: sip:+17083957689@fl.gg;tag=7af9c92d-6948-4b21-a291-d9299cec4d1e
Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK07a8.05ac027122acb1a9586ab68302f2f7b3.0
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK07a8.2a5b9924d9905299fab9c996a5d197ec.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB52e76493694f73f0
Call-ID: 221000176_79612347@206.147.88.72
CSeq: 3003 ACK
Content-Length: 0
-- <PJSIP/provider-00000018> Playing 'vm-intro.ulaw' (language 'en')
-- <PJSIP/provider-00000018> Playing 'beep.ulaw' (language 'en')
-- Recording the message
<— Received SIP request (599 bytes) from UDP:34.226.36.35:5060 —>
BYE sip:24.13.121.136:5060 SIP/2.0
Record-Route: sip:34.226.36.35;lr
Max-Forwards: 67
Record-Route: sip:34.211.73.216;lr
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
To: sip:+17083957689@fl.gg;tag=7af9c92d-6948-4b21-a291-d9299cec4d1e
Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bKd6a8.4cf6377ac2154f670510d3a9609f8062.0
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bKd6a8.8239abb6a8085afcf73315f3168af997.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB54e08be5694f73f0
Call-ID: 221000176_79612347@206.147.88.72
CSeq: 3004 BYE
Content-Length: 0
<— Transmitting SIP response (622 bytes) to UDP:34.226.36.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bKd6a8.4cf6377ac2154f670510d3a9609f8062.0
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bKd6a8.8239abb6a8085afcf73315f3168af997.0
Via: SIP/2.0/UDP 206.147.88.72:5060;branch=z9hG4bK0cB54e08be5694f73f0
Record-Route: sip:34.226.36.35;lr
Record-Route: sip:34.211.73.216;lr
Call-ID: 221000176_79612347@206.147.88.72
From: “[V]JEFFERSON TI” sip:+17086326111@fl.gg;tag=gK0c6c21b4
To: sip:+17083957689@fl.gg;tag=7af9c92d-6948-4b21-a291-d9299cec4d1e
CSeq: 3004 BYE
Server: Asterisk PBX 22.4.1
Content-Length: 0
-- User hung up
-- Recording was 0 seconds long but needs to be at least 3 - abandoning
== Spawn extension (phones, 200, 3) exited non-zero on ‘PJSIP/provider-00000018’
<— Received SIP request (982 bytes) from UDP:10.0.0.2:44889 —>
REGISTER sip:192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:44889;branch=z9hG4bK-524287-1—b0ee3ea78a6c8834;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.2:44889;rinstance=0ee7ca26fffafcc8;transport=UDP
To: sip:enigma@192.168.1.2;transport=UDP
From: sip:enigma@192.168.1.2;transport=UDP;tag=fe671e77
Call-ID: ZYjC83uyAB4wqcPm7QTwAA..
CSeq: 29 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
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=“1750114399/664144158da417d0f1fed86bab730150”,uri=“sip:192.168.1.2;transport=UDP”,response=“dc4d4eda55835bf76de800506ea1dcb8”,cnonce=“34280cac213bc7d424cc55e48b00ad5a”,nc=00000002,qop=auth,algorithm=MD5,opaque=“43125e15238d97fb”
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0
<— Transmitting SIP response (509 bytes) to UDP:10.0.0.2:44889 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.2:44889;rport=44889;received=10.0.0.2;branch=z9hG4bK-524287-1—b0ee3ea78a6c8834
Call-ID: ZYjC83uyAB4wqcPm7QTwAA..
From: sip:enigma@192.168.1.2;tag=fe671e77
To: sip:enigma@192.168.1.2;tag=z9hG4bK-524287-1—b0ee3ea78a6c8834
CSeq: 29 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1750114452/4eb117cd4be854f797ae96089c03ea66”,opaque=“147b578b743e3783”,stale=true,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.4.1
Content-Length: 0
<— Received SIP request (982 bytes) from UDP:10.0.0.2:44889 —>
REGISTER sip:192.168.1.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:44889;branch=z9hG4bK-524287-1—07ce87b83309a68c;rport
Max-Forwards: 70
Contact: sip:enigma@10.0.0.2:44889;rinstance=0ee7ca26fffafcc8;transport=UDP
To: sip:enigma@192.168.1.2;transport=UDP
From: sip:enigma@192.168.1.2;transport=UDP;tag=fe671e77
Call-ID: ZYjC83uyAB4wqcPm7QTwAA..
CSeq: 30 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
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=“1750114452/4eb117cd4be854f797ae96089c03ea66”,uri=“sip:192.168.1.2;transport=UDP”,response=“76495a7e3dbf9723977c6c714d34ff98”,cnonce=“f5d25226a43b2cbaf64bde128966233d”,nc=00000001,qop=auth,algorithm=MD5,opaque=“147b578b743e3783”
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0
<— Transmitting SIP response (482 bytes) to UDP:10.0.0.2:44889 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:44889;rport=44889;received=10.0.0.2;branch=z9hG4bK-524287-1—07ce87b83309a68c
Call-ID: ZYjC83uyAB4wqcPm7QTwAA..
From: sip:enigma@192.168.1.2;tag=fe671e77
To: sip:enigma@192.168.1.2;tag=z9hG4bK-524287-1—07ce87b83309a68c
CSeq: 30 REGISTER
Date: Mon, 16 Jun 2025 22:54:12 GMT
Contact: sip:enigma@10.0.0.2:44889;transport=UDP;rinstance=0ee7ca26fffafcc8;expires=59
Expires: 60
Server: Asterisk PBX 22.4.1
Content-Length: 0