Hi, using Asterisk 12 i can’t communicate with peers using the websocket dialing from ws to sip works as expected, dialing from sip to ws can’t locate the peer and dialing from ws to ws is failing too.
Here is the dialplan:
[ Context 'phones' created by 'pbx_config' ]
'_15XXX' => 1. Dial(PJSIP/${EXTEN:1}) [pbx_config]
2. Hangup() [pbx_config]
'_5XXX' => 1. Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) [pbx_config]
2. Hangup() [pbx_config]
The Contact List of the peers:
pjsip show endpoint 5001
Endpoint 5001:
AOR 5001:
Contact sip:5001@10.0.1.102:48116;transport=ws;rtcweb-breaker=no:
available = yes
RTT = 114700 microseconds
pjsip show endpoint 5000
Endpoint 5000:
AOR 5000:
Contact sip:5000@10.0.1.110:5063:
available = yes
RTT = 414624 microseconds
The result of dialing from SIP to WS
[code] – Executing [5001@phones:1] Dial(“PJSIP/5000-00000002”, “PJSIP/5001/sip:5001@10.0.1.102:48116;transport=ws;rtcweb-breaker=no”) in new stack
[Dec 19 12:28:53] WARNING[3492][C-00000002]: app_dial.c:2423 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 [5001@phones:2] Hangup(“PJSIP/5000-00000002”, “”) in new stack
== Spawn extension (phones, 5001, 2) exited non-zero on ‘PJSIP/5000-00000002’
– Executing [15001@phones:1] Dial(“PJSIP/5000-00000003”, “PJSIP/5001”) in new stack
[Dec 19 12:29:12] WARNING[3493][C-00000003]: app_dial.c:2423 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 [15001@phones:2] Hangup(“PJSIP/5000-00000003”, “”) in new stack
== Spawn extension (phones, 15001, 2) exited non-zero on ‘PJSIP/5000-00000003’
[/code]
The PJSIP debug:
[code]PJSIP Logging enabled
<— Received SIP request (926 bytes) from UDP:10.0.1.110:5063 —>
INVITE sip:5001@10.0.1.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.110:5063;branch=z9hG4bK484473620
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102
Call-ID: 375194722@10.0.1.110
CSeq: 1 INVITE
Contact: sip:5000@10.0.1.110:5063
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.70.23.2 00:15:65:39:02:92
Supported: replaces,100rel
P-Early-Media: supported
Expires: 360
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 302
v=0
o=- 20049 20049 IN IP4 10.0.1.110
s=SDP data
c=IN IP4 10.0.1.110
t=0 0
m=audio 11796 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<— Transmitting SIP response (419 bytes) to UDP:10.0.1.110:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.110:5063;received=10.0.1.110;branch=z9hG4bK484473620
Call-ID: 375194722@10.0.1.110
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102;tag=z9hG4bK484473620
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1387477794/78c54cbbd8af3528f2fb1cb70d78cf1c”,opaque=“4fcb5c3a01b6d877”,algorithm=md5,qop="auth"
Content-Length: 0
<— Received SIP request (255 bytes) from UDP:10.0.1.110:5063 —>
ACK sip:5001@10.0.1.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.110:5063;branch=z9hG4bK484473620
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102;tag=z9hG4bK484473620
Call-ID: 375194722@10.0.1.110
CSeq: 1 ACK
Content-Length: 0
<— Received SIP request (1192 bytes) from UDP:10.0.1.110:5063 —>
INVITE sip:5001@10.0.1.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.110:5063;branch=z9hG4bK388011450
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102
Call-ID: 375194722@10.0.1.110
CSeq: 2 INVITE
Contact: sip:5000@10.0.1.110:5063
Authorization: Digest username=“5000”, realm=“asterisk”, nonce=“1387477794/78c54cbbd8af3528f2fb1cb70d78cf1c”, uri="sip:5001@10.0.1.102", response=“ee80bdc04900a490555fa91c3b79abba”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“4fcb5c3a01b6d877”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.70.23.2 00:15:65:39:02:92
Supported: replaces,100rel
P-Early-Media: supported
Expires: 360
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 302
v=0
o=- 20049 20049 IN IP4 10.0.1.110
s=SDP data
c=IN IP4 10.0.1.110
t=0 0
m=audio 11796 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<— Transmitting SIP response (245 bytes) to UDP:10.0.1.110:5063 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.110:5063;received=10.0.1.110;branch=z9hG4bK388011450
Call-ID: 375194722@10.0.1.110
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102
CSeq: 2 INVITE
Content-Length: 0
<— Received SIP request (1192 bytes) from UDP:10.0.1.110:5063 —>
INVITE sip:5001@10.0.1.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.110:5063;branch=z9hG4bK388011450
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102
Call-ID: 375194722@10.0.1.110
CSeq: 2 INVITE
Contact: sip:5000@10.0.1.110:5063
Authorization: Digest username=“5000”, realm=“asterisk”, nonce=“1387477794/78c54cbbd8af3528f2fb1cb70d78cf1c”, uri="sip:5001@10.0.1.102", response=“ee80bdc04900a490555fa91c3b79abba”, algorithm=MD5, cnonce=“0a4f113b”, opaque=“4fcb5c3a01b6d877”, qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.70.23.2 00:15:65:39:02:92
Supported: replaces,100rel
P-Early-Media: supported
Expires: 360
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 302
v=0
o=- 20049 20049 IN IP4 10.0.1.110
s=SDP data
c=IN IP4 10.0.1.110
t=0 0
m=audio 11796 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
-- Executing [5001@phones:1] Dial("PJSIP/5000-00000004", "PJSIP/5001/sip:5001@10.0.1.102:48116;transport=ws;rtcweb-breaker=no") in new stack
<— Transmitting SIP response (245 bytes) to UDP:10.0.1.110:5063 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.110:5063;received=10.0.1.110;branch=z9hG4bK388011450
Call-ID: 375194722@10.0.1.110
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102
CSeq: 2 INVITE
Content-Length: 0
[Dec 19 12:29:55] WARNING[3503][C-00000004]: app_dial.c:2423 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 [5001@phones:2] Hangup(“PJSIP/5000-00000004”, “”) in new stack
== Spawn extension (phones, 5001, 2) exited non-zero on ‘PJSIP/5000-00000004’
<— Transmitting SIP response (312 bytes) to UDP:10.0.1.110:5063 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.1.110:5063;received=10.0.1.110;branch=z9hG4bK388011450
Call-ID: 375194722@10.0.1.110
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102;tag=a3b1b278-f170-45e0-9303-c32f561156e3
CSeq: 2 INVITE
Reason: Q.850;cause=3
Content-Length: 0
<— Received SIP request (275 bytes) from UDP:10.0.1.110:5063 —>
ACK sip:5001@10.0.1.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.110:5063;branch=z9hG4bK388011450
From: “test” sip:5000@10.0.1.102;tag=1576769793
To: sip:5001@10.0.1.102;tag=a3b1b278-f170-45e0-9303-c32f561156e3
Call-ID: 375194722@10.0.1.110
CSeq: 2 ACK
Content-Length: 0
Airo-Shaka*CLI> pjsip set logger off
PJSIP Logging disabled
[/code]
Best Regards.