Hello, as I shared you before, I am trying to connect two asterisk servers using PJSIP trunk.
My goal is: When call arrives in server1 to map the call/forward the call in to the server2 using PJSIP trunk between the two servers.
Note: All users and queues calls are in server2. Server1 will just recieve the signal from the SIP trunk/E1 PSTN.
Current, The incoming call is working. The call arrives in server2.
The issue is Outbound calls.
Here is My Trunk between the two asterisk servers
Server1: 192.x.x…x.76
Server2: 192.x.x.x75
server1:in psjip.conf
[mytrunk]
type=registration
outbound_auth=mytrunk
server_uri=sip:192.x.x.x75
client_uri=sip:user@192.x.x.x75
retry_interval=60
[mytrunk]
type=auth
auth_type=userpass
password=user
username=2233444
[mytrunk]
type=aor
contact=sip:192.x.x.x75:5060
[mytrunk]
type=endpoint
context=from-pstn
disallow=all
allow=ulaw
outbound_auth=mytrunk
aors=mytrunk
[mytrunk]
type=identify
endpoint=mytrunk
match=192.x.x.75
server2 in pjsip.conf
[mytrunk]
type=registration
outbound_auth=mytrunk
server_uri=sip:192.x.x.x75
client_uri=sip:user@192.x.x.x75
retry_interval=60
[mytrunk]
type=auth
auth_type=userpass
password=user
username=2233444
[mytrunk]
type=aor
contact=sip:192.x.x.x75:5060
[mytrunk]
type=endpoint
context=from-pstn
disallow=all
allow=ulaw
outbound_auth=mytrunk
aors=mytrunk
[mytrunk]
type=identify
endpoint=mytrunk
match=192.168.x.x.76
when I am making ouboud call from server2 I am getting this error from the PJSIP logger on:
PJSIP Logging enabled
<— Received SIP request (983 bytes) from UDP:192.x.x.x.x:55597 —>
INVITE sip:09xxxxxxxx@192.x.x.x75 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport;branch=z9hG4bKPj2fdbf4175dbc411abba76811f343beeb
Max-Forwards: 70
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxxxx@192.x.x.x75
Contact: sip:3001@192.x.x.x.x:55597;ob
Call-ID: a2a41e0b469346aba0bddff97503e1bb
CSeq: 5245 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: SoftphonePro 5.3.0.0
Content-Type: application/sdp
Content-Length: 345
v=0
o=- 3906265364 3906265364 IN IP4 192.x.x.x.x
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 65206 RTP/AVP 8 0 101
c=IN IP4 192.x.x.x.x
b=TIAS:64000
a=rtcp:65207 IN IP4 192.x.x.x.x
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1397239416 cname:0855010c36a87d3e
<— Transmitting SIP response (559 bytes) to UDP:192.x.x.x.x:55597 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport=55597;received=192.x.x.x.x;branch=z9hG4bKPj2fdbf4175dbc411abba76811f343beeb
Call-ID: a2a41e0b469346aba0bddff97503e1bb
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxxxx@192.x.x.x75;tag=z9hG4bKPj2fdbf4175dbc411abba76811f343beeb
CSeq: 5245 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1697265752/a05feff2edbd96306ac26ed877d255e1”,opaque=“01bfb4e1354418e8”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 18.19.0
Content-Length: 0
<— Received SIP request (385 bytes) from UDP:192.x.x.x.x:55597 —>
ACK sip:09xxxxx@192.x.x.x75 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport;branch=z9hG4bKPj2fdbf4175dbc411abba76811f343beeb
Max-Forwards: 70
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: <sip:09xxxxxx
@192.x.x.x75>;tag=z9hG4bKPj2fdbf4175dbc411abba76811f343beeb
Call-ID: a2a41e0b469346aba0bddff97503e1bb
CSeq: 5245 ACK
Content-Length: 0
<— Received SIP request (1282 bytes) from UDP:192.x.x.x.x:55597 —>
INVITE sip:09xxxxxx@192.x.x.x75 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport;branch=z9hG4bKPj77b7929478aa4dc08a0e5bfe49969ad1
Max-Forwards: 70
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxxxx@192.x.x.x75
Contact: sip:3001@192.x.x.x.x:55597;ob
Call-ID: a2a41e0b469346aba0bddff97503e1bb
CSeq: 5246 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: SoftphonePro 5.3.0.0
Authorization: Digest username=“3001”, realm=“asterisk”, nonce=“1697265752/a05feff2edbd96306ac26ed877d255e1”, uri=“sip:09xxxx@192.xxx.75”, response=“5f234cc367b18341ec3febcb85736787”, algorithm=MD5, cnonce=“54946f8f648f4367811ce679dc92ea7e”, opaque=“01bfb4e1354418e8”, qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 345
v=0
o=- 3906265364 3906265364 IN IP4 192.x.x.x.x
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 65206 RTP/AVP 8 0 101
c=IN IP4 192.x.x.x.x
b=TIAS:64000
a=rtcp:65207 IN IP4 192.x.x.x.x
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1397239416 cname:0855010c36a87d3e
<— Transmitting SIP response (361 bytes) to UDP:192.x.x.x.x:55597 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport=55597;received=192.x.x.x.x;branch=z9hG4bKPj77b7929478aa4dc08a0e5bfe49969ad1
Call-ID: a2a41e0b469346aba0bddff97503e1bb
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxx@192.x.x.x75
CSeq: 5246 INVITE
Server: Asterisk PBX 18.19.0
Content-Length: 0
-- Executing [09xxxxx@mytrunk:1] Set("PJSIP/3001-00000008", "CALLERID(number)=+26xxxxxxx20") in new stack
-- Executing [09xxxxxxx@mytrunk:2] Dial("PJSIP/3001-00000008", "PJSIP/09xxxxx@mytrunk") in new stack
-- Called PJSIP/09xxxxxx@mytrunk
<— Transmitting SIP request (923 bytes) to UDP:192.x.x.x76:5060 —>
INVITE sip:09xxxxx@192.x.x.x76:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x75:5060;rport;branch=z9hG4bKPj506f03e8-4b0a-4b31-a823-7b8233fe03d6
From: sip:+26xxxxxxx20@192.x.x.x75;tag=da2401e1-bdf9-46f4-98ad-5b9c4d21a931
To: sip:09xxxxx@192.x.x.x76
Contact: sip:asterisk@192.x.x.x75:5060
Call-ID: 458f338c-6537-4fdf-a9f5-5f2652304eca
CSeq: 3158 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.19.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 177994976 177994976 IN IP4 192.x.x.x75
s=Asterisk
c=IN IP4 192.x.x.x75
t=0 0
m=audio 13290 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Received SIP response (424 bytes) from UDP:192.x.x.x76:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.x.x.x75:5060;rport=5060;received=192.x.x.x75;branch=z9hG4bKPj506f03e8-4b0a-4b31-a823-7b8233fe03d6
Call-ID: 458f338c-6537-4fdf-a9f5-5f2652304eca
From: sip:+26xxxxxxx20@192.x.x.x75;tag=da2401e1-bdf9-46f4-98ad-5b9c4d21a931
To: sip:09xxxxx@192.x.x.x76;tag=8c4dcb9b-865a-43e4-b8c8-c856c3ea4909
CSeq: 3158 INVITE
Server: Asterisk PBX 18.19.0
Content-Length: 0
<— Transmitting SIP request (439 bytes) to UDP:192.x.x.x76:5060 —>
ACK sip:09xxxx@192.xxx.76:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x75:5060;rport;branch=z9hG4bKPj506f03e8-4b0a-4b31-a823-7b8233fe03d6
From: sip:+26xxxxxxx20@192.x.x.x75;tag=da2401e1-bdf9-46f4-98ad-5b9c4d21a931
To: sip:09xxxxx@192.x.x.x76;tag=8c4dcb9b-865a-43e4-b8c8-c856c3ea4909
Call-ID: 458f338c-6537-4fdf-a9f5-5f2652304eca
CSeq: 3158 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.19.0
Content-Length: 0
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [09xxxxxxx
@mytrunk:3] Hangup(“PJSIP/3001-00000008”, “”) in new stack
== Spawn extension (mytrunk, 09xxxxxx, 3) exited non-zero on ‘PJSIP/3001-00000008’
<— Transmitting SIP response (428 bytes) to UDP:192.x.x.x.x:55597 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport=55597;received=192.x.x.x.x;branch=z9hG4bKPj77b7929478aa4dc08a0e5bfe49969ad1
Call-ID: a2a41e0b469346aba0bddff97503e1bb
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxxx@192.x.x.x75;tag=f124ae08-e3e5-4491-80a0-c4fa71e1af05
CSeq: 5246 INVITE
Server: Asterisk PBX 18.19.0
Reason: Q.850;cause=1
Content-Length: 0
<— Received SIP request (380 bytes) from UDP:192.x.x.x.x:55597 —>
ACK sip:09xxxxxx@192.x.x.x75 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x.x:55597;rport;branch=z9hG4bKPj77b7929478aa4dc08a0e5bfe49969ad1
Max-Forwards: 70
From: sip:3001@192.x.x.x75;tag=213f9302c3c9496989dcd16e24e93248
To: sip:09xxxxxx@192.x.x.x75;tag=f124ae08-e3e5-4491-80a0-c4fa71e1af05
Call-ID: a2a41e0b469346aba0bddff97503e1bb
CSeq: 5246 ACK
Content-Length: 0