As requested the content of pjsip.conf (1st server):
; GLOBAL
[global]
debug=no
[transport-udp]
type=transport
protocol=udp
bind=192.168.87.11:5060
local_net=192.168.0.0/16
[astp4]
type=registration
transport=transport-udp
server_uri=sip:192.168.87.11:5060
client_uri=sip:astp4@192.168.89.1:5060
[astp4]
type=endpoint
transport=transport-udp
context=internal
direct_media=no
disallow=all
allow=alaw
aors=astp4
rewrite_contact=yes
[astp4]
type=aor
qualify_frequency=30
max_contacts=5
[astp4]
type=identify
endpoint=astp4
match=192.168.89.1
match=192.168.87.11
[astp4]
type=contact
uri=sip:192.168.89.1:5060
[transport-udp-out]
type=transport
protocol=udp
bind=195.162.X.X:5060
local_net=195.162.X.X/29
[outtrunk]
type=endpoint
transport=transport-udp-out
context=internal
allow=alaw
aors=outtrunk
[outtrunk]
type=aor
qualify_frequency=30
max_contacts=5
contact=sip:195.162.X.X:5060
[outtrunk]
type=identify
endpoint=outtrunk
match=195.162.X.X
match=195.162.X.X
[outtrunk]
type=contact
uri=sip:195.162.X.X:5060
[42680]
type=aor
max_contacts=5
[42680]
type=endpoint
context=internal
disallow=all
allow=alaw
auth=auth42680
aors=42680
[auth42680]
type=auth
auth_type=md5
md5_cred=c1faea51dfbaf89c2616c8269a47f0da
username=42680
[41040]
type=aor
max_contacts=5
[41040]
type=endpoint
context=internal
disallow=all
allow=alaw
auth=auth41040
aors=41040
[auth41040]
type=auth
auth_type=md5
md5_cred=
username=41040
extensions.conf of 1st server:
[code][general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[internal]
; External TEST
exten => 93985010,1,Verbose(Callerid is ${CALLERID(number)})
same => n,dial(PJSIP/astp4/sip:X@192.168.89.1:5060)
same => n,Hangup()
; Internal TEST
exten => _41040,1,dial(PJSIP/41040,10)[/code]
pjsip.conf of 2nd server:
[code]; GLOBAL
[global]
debug=no
[transport-udp]
type=transport
protocol=udp
bind=192.168.89.1:5060
local_net=192.168.0.0/16
[transport-udp-out]
type=transport
protocol=udp
bind=195.162.X.X:5060
local_net=195.162.X.X/29
[astp3]
type=registration
transport=transport-udp
server_uri=sip:192.168.89.1:5060
client_uri=sip:astp3@192.168.87.11:5060
[astp3]
type=endpoint
transport=transport-udp
context=internal
direct_media=no
disallow=all
allow=alaw
aors=astp3
rewrite_contact=yes
[astp3]
type=aor
qualify_frequency=30
max_contacts=5
[astp3]
type=identify
endpoint=astp3
match=192.168.87.11
match=192.168.89.1
[astp3]
type=contact
uri=sip:192.168.87.11:5060
[outtrunk]
type=endpoint
transport=transport-udp-out
context=internal
allow=alaw
aors=outtrunk
[outtrunk]
type=aor
qualify_frequency=30
max_contacts=5
contact=sip:195.162.X.X:5060
[outtrunk]
type=identify
endpoint=outtrunk
match=195.162.X.X
match=195.162.X.X
[outtrunk]
type=contact
uri=sip:195.162.X.X:5060
[42680]
type=aor
max_contacts=5
[42680]
type=endpoint
context=internal
disallow=all
allow=alaw
auth=auth42680
aors=42680
[auth42680]
type=auth
auth_type=md5
md5_cred=
username=42680
[41040]
type=aor
max_contacts=5
[41040]
type=endpoint
context=internal
disallow=all
allow=alaw
auth=auth41040
aors=41040
[auth41040]
type=auth
auth_type=md5
md5_cred=
username=41040
[/code]
extensions.conf of 2nd server:
[code][general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[via-out]
exten => _939850XX,1,Verbose(Call via outtrunk van ${EXTEN}.)
same => n,Gosub(internal,${EXTEN},1)
[internal]
; Internal TEST
exten => 41040,1,Verbose(Callerid is ${CALLERID(number)})
same => n,dial(PJSIP/astp3/sip:41040@192.168.87.11:5060)
; External TEST
exten => 93985010,1,Verbose(Callerid is ${CALLERID(number)})
same => n,answer()
same => n,playback(demo-congrats)
same => n,hangup()[/code]
Internal call between 42268 and 41104 (as above, which fails)
logger of 1st server:
[code]astp3*CLI> pjsip set logger on
PJSIP Logging enabled
<— Transmitting SIP request (433 bytes) to UDP:195.162.X.X:5060 —>
OPTIONS sip:195.162.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 195.162.X.X:5060;rport;branch=z9hG4bKPj19db1586-ec33-42ef-a512-b89fc8254369
From: sip:asterisk@195.162.X.X;tag=9de71359-6536-4d17-977c-6e13d1ed69cf
To: sip:195.162.X.X
Contact: sip:asterisk@195.162.X.X:5060
Call-ID: 239f576b-8ef8-4877-9fe3-4b9fa9898fbf
CSeq: 52721 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Received SIP response (586 bytes) from UDP:195.162.X.X:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.162.X.X:5060;branch=z9hG4bKPj19db1586-ec33-42ef-a512-b89fc8254369;rport=5060
From: sip:asterisk@195.162.X.X;tag=9de71359-6536-4d17-977c-6e13d1ed69cf
To: sip:195.162.X.X
Call-ID: 239f576b-8ef8-4877-9fe3-4b9fa9898fbf
CSeq: 52721 OPTIONS
Contact: sip:ANONYMOUS@195.162.X.X:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Supported: timer,100rel,replaces
Content-Length: 0
<— Received SIP request (897 bytes) from UDP:192.168.89.1:5060 —>
INVITE sip:41040@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.89.1:5060;rport;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11
Contact: sip:asterisk@192.168.89.1:5060
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
CSeq: 2619 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 235
v=0
o=- 343162209 343162209 IN IP4 192.168.89.1
s=Asterisk
c=IN IP4 192.168.89.1
t=0 0
m=audio 10416 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP response (371 bytes) to UDP:192.168.89.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.89.1:5060;rport=5060;received=192.168.89.1;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11
CSeq: 2619 INVITE
Server: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP request (429 bytes) to UDP:192.168.87.11:5060 —>
OPTIONS sip:s@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
From: sip:asterisk@195.162.X.X;tag=ba0ef886-be48-40a0-9684-c94036607eee
To: sip:s@192.168.87.11
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: e49296f6-52f0-44c1-99b3-a5a48c5a2c92
CSeq: 41921 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Received SIP request (429 bytes) from UDP:195.162.X.X:5060 —>
OPTIONS sip:s@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
From: sip:asterisk@195.162.X.X;tag=ba0ef886-be48-40a0-9684-c94036607eee
To: sip:s@192.168.87.11
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: e49296f6-52f0-44c1-99b3-a5a48c5a2c92
CSeq: 41921 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP response (911 bytes) to UDP:195.162.X.X:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.87.11:5060;rport=5060;received=195.162.X.X;branch=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
Call-ID: e49296f6-52f0-44c1-99b3-a5a48c5a2c92
From: sip:asterisk@195.162.X.X;tag=ba0ef886-be48-40a0-9684-c94036607eee
To: sip:s@192.168.87.11;tag=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
CSeq: 41921 OPTIONS
Accept: application/sdp, application/dialog-info+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX 13.6.0
Content-Length: 0
<— Received SIP response (911 bytes) from UDP:192.168.87.11:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.87.11:5060;rport=5060;received=195.162.X.X;branch=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
Call-ID: e49296f6-52f0-44c1-99b3-a5a48c5a2c92
From: sip:asterisk@195.162.X.X;tag=ba0ef886-be48-40a0-9684-c94036607eee
To: sip:s@192.168.87.11;tag=z9hG4bKPjed994413-eab3-42f9-9e1b-bc09f8575312
CSeq: 41921 OPTIONS
Accept: application/sdp, application/dialog-info+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Transmitting SIP response (436 bytes) to UDP:192.168.89.1:5060 —>
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.89.1:5060;rport=5060;received=192.168.89.1;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11;tag=6bafac6f-5432-4bc6-b8ef-513a85ca1115
CSeq: 2619 INVITE
Server: Asterisk PBX 13.6.0
Reason: Q.850;cause=0
Content-Length: 0
<— Received SIP request (426 bytes) from UDP:192.168.89.1:5060 —>
ACK sip:41040@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.89.1:5060;rport;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11;tag=6bafac6f-5432-4bc6-b8ef-513a85ca1115
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
CSeq: 2619 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP request (907 bytes) to UDP:172.17.105.102:5062 —>
INVITE sip:41040@172.17.105.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.87.11:5060;rport;branch=z9hG4bKPjefd66edb-d39d-49f0-ae1c-b79075e257ff
From: “42680” sip:42680@195.162.X.X;tag=af72ecb9-442f-47a6-9424-dd57cc46d7ea
To: sip:41040@172.17.105.102
Contact: sip:asterisk@192.168.87.11:5060
Call-ID: cb814e4c-4ae1-497c-9dcf-309485df58f8
CSeq: 11902 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 583048194 583048194 IN IP4 192.168.87.11
s=Asterisk
c=IN IP4 192.168.87.11
t=0 0
m=audio 13988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
astp3*CLI>
Disconnected from Asterisk server
[/code]
logger of 2nd server:
[code]astp4*CLI> pjsip set logger on
PJSIP Logging enabled
<— Received SIP request (1077 bytes) from UDP:172.17.107.67:5060 —>
INVITE sip:41040@astp4.internal.xyz SIP/2.0
Via: SIP/2.0/UDP 172.17.107.67:5060;branch=z9hG4bK1802527485;rport
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
CSeq: 440 INVITE
Contact: “42680” sip:42680@172.17.107.67:5060
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.46
Privacy: none
P-Preferred-Identity: “42680” sip:42680@astp4.internal.xyz
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 356
v=0
o=42680 8000 8000 IN IP4 172.17.107.67
s=SIP Call
c=IN IP4 172.17.107.67
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<— Transmitting SIP response (511 bytes) to UDP:172.17.107.67:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.17.107.67:5060;rport=5060;received=172.17.107.67;branch=z9hG4bK1802527485
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz;tag=z9hG4bK1802527485
CSeq: 440 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1450787925/4e0b3993127aeb702c01841b1273c2a8”,opaque=“216bbc505d1f5e42”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.6.0
Content-Length: 0
<— Received SIP request (325 bytes) from UDP:172.17.107.67:5060 —>
ACK sip:41040@astp4.internal.xyz SIP/2.0
Via: SIP/2.0/UDP 172.17.107.67:5060;branch=z9hG4bK1802527485;rport
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz;tag=z9hG4bK1802527485
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
CSeq: 440 ACK
Content-Length: 0
<— Received SIP request (1359 bytes) from UDP:172.17.107.67:5060 —>
INVITE sip:41040@astp4.internal.xyz SIP/2.0
Via: SIP/2.0/UDP 172.17.107.67:5060;branch=z9hG4bK1721025199;rport
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
CSeq: 441 INVITE
Contact: “42680” sip:42680@172.17.107.67:5060
Authorization: Digest username=“42680”, realm=“asterisk”, nonce=“1450787925/4e0b3993127aeb702c01841b1273c2a8”, uri=“sip:41040@astp4.internal.xyz”, response=“7168835c38e7a7e7f4dbd311c21a97b9”, algorithm=md5, cnonce=“14144175”, opaque=“216bbc505d1f5e42”, qop=auth, nc=00000002
Max-Forwards: 70
User-Agent: Grandstream GXV3240 1.0.3.46
Privacy: none
P-Preferred-Identity: “42680” sip:42680@astp4.internal.xyz
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 356
v=0
o=42680 8000 8000 IN IP4 172.17.107.67
s=SIP Call
c=IN IP4 172.17.107.67
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<— Transmitting SIP response (336 bytes) to UDP:172.17.107.67:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.107.67:5060;rport=5060;received=172.17.107.67;branch=z9hG4bK1721025199
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz
CSeq: 441 INVITE
Server: Asterisk PBX 13.6.0
Content-Length: 0
Callerid is 42680
<— Transmitting SIP request (897 bytes) to UDP:192.168.87.11:5060 —>
INVITE sip:41040@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.89.1:5060;rport;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11
Contact: sip:asterisk@192.168.89.1:5060
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
CSeq: 2619 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Type: application/sdp
Content-Length: 235
v=0
o=- 343162209 343162209 IN IP4 192.168.89.1
s=Asterisk
c=IN IP4 192.168.89.1
t=0 0
m=audio 10416 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Received SIP response (371 bytes) from UDP:192.168.87.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.89.1:5060;rport=5060;received=192.168.89.1;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11
CSeq: 2619 INVITE
Server: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP request (432 bytes) to UDP:195.162.X.X:5060 —>
OPTIONS sip:195.162.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 195.162.X.X:5060;rport;branch=z9hG4bKPj34e71d17-504d-43e4-b7de-9a0653905c4f
From: sip:asterisk@195.162.X.X;tag=3901a13a-8815-417e-a1ed-6144cc865b56
To: sip:195.162.X.X
Contact: sip:asterisk@195.162.X.X:5060
Call-ID: 957f0c3d-5409-4c56-b303-fe7f8e6477c2
CSeq: 1261 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP request (432 bytes) to UDP:195.162.X.X:5060 —>
OPTIONS sip:195.162.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 195.162.X.X:5060;rport;branch=z9hG4bKPj34e71d17-504d-43e4-b7de-9a0653905c4f
From: sip:asterisk@195.162.X.X;tag=3901a13a-8815-417e-a1ed-6144cc865b56
To: sip:195.162.X.X
Contact: sip:asterisk@195.162.X.X:5060
Call-ID: 957f0c3d-5409-4c56-b303-fe7f8e6477c2
CSeq: 1261 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Received SIP response (585 bytes) from UDP:195.162.X.X:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.162.X.X:5060;branch=z9hG4bKPj34e71d17-504d-43e4-b7de-9a0653905c4f;rport=5060
From: sip:asterisk@195.162.X.X;tag=3901a13a-8815-417e-a1ed-6144cc865b56
To: sip:195.162.X.X
Call-ID: 957f0c3d-5409-4c56-b303-fe7f8e6477c2
CSeq: 1261 OPTIONS
Contact: sip:ANONYMOUS@195.162.X.X:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Supported: timer,100rel,replaces
Content-Length: 0
<— Received SIP response (585 bytes) from UDP:195.162.X.X:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.162.X.X:5060;branch=z9hG4bKPj34e71d17-504d-43e4-b7de-9a0653905c4f;rport=5060
From: sip:asterisk@195.162.X.X;tag=3901a13a-8815-417e-a1ed-6144cc865b56
To: sip:195.162.X.X
Call-ID: 957f0c3d-5409-4c56-b303-fe7f8e6477c2
CSeq: 1261 OPTIONS
Contact: sip:ANONYMOUS@195.162.X.X:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Supported: timer,100rel,replaces
Content-Length: 0
<— Received SIP response (436 bytes) from UDP:192.168.87.11:5060 —>
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.89.1:5060;rport=5060;received=192.168.89.1;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11;tag=6bafac6f-5432-4bc6-b8ef-513a85ca1115
CSeq: 2619 INVITE
Server: Asterisk PBX 13.6.0
Reason: Q.850;cause=0
Content-Length: 0
<— Transmitting SIP request (426 bytes) to UDP:192.168.87.11:5060 —>
ACK sip:41040@192.168.87.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.89.1:5060;rport;branch=z9hG4bKPj679978e7-6540-4948-b80b-2847af0be15e
From: “42680” sip:42680@192.168.89.1;tag=87e98043-a980-4501-ab3c-c7d411e32224
To: sip:41040@192.168.87.11;tag=6bafac6f-5432-4bc6-b8ef-513a85ca1115
Call-ID: 190bb3f3-912d-4719-9da1-e7c9a369fa88
CSeq: 2619 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0
<— Transmitting SIP response (414 bytes) to UDP:172.17.107.67:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.17.107.67:5060;rport=5060;received=172.17.107.67;branch=z9hG4bK1721025199
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.xyz;tag=b5eb7e81-29ba-4e17-9239-268d00057f6a
CSeq: 441 INVITE
Server: Asterisk PBX 13.6.0
Reason: Q.850;cause=34
Content-Length: 0
<— Received SIP request (344 bytes) from UDP:172.17.107.67:5060 —>
ACK sip:41040@astp4.internal.xyz SIP/2.0
Via: SIP/2.0/UDP 172.17.107.67:5060;branch=z9hG4bK1721025199;rport
From: “42680” sip:42680@astp4.internal.xyz;tag=992053134
To: sip:41040@astp4.internal.x;tag=b5eb7e81-29ba-4e17-9239-268d00057f6a
Call-ID: 1485638553-5060-45@BHC.BH.BAH.GH
CSeq: 441 ACK
Content-Length: 0
astp4*CLI>
Disconnected from Asterisk server[/code]