Hi,
When we dial an extension, some times the request is getting timed-out and not connecting to the destination. This is happening very frequently. But some times it connect and ring successfully as expected.
Using - “Asterisk 20.5” version
SIP Softphone app - “Telephone”
Please help to identify or resolve this issue.
You would need to show an actual trace from “pjsip set logger on” of a call attempt including Asterisk console output.
This is the call log, with a similar scenario like, even if we attended the call at the destination (extn: 201) side, call is still ringing in source side (extn : 200). ie, not establishing the connection.
(here we tried to call from extn 200 to extn 201)
Please help
-- Executing [201@internal:1] Verbose("PJSIP/200-00000002", "call was placed..)") in new stack
call was placed..)
-- Executing [201@internal:2] Dial("PJSIP/200-00000002", "PJSIP/201,60)") in new stack
<--- Transmitting SIP request (922 bytes) to UDP:201.202.203.204:5060 --->
INVITE sip:201@201.202.203.204:5060;ob SIP/2.0
Via: SIP/2.0/UDP 101.102.103.104:5060;rport;branch=z9hG4bKPj92692640-20ec-4fac-a8e8-44cc006be541
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204;ob>
Contact: <sip:asterisk@101.102.103.104:5060>
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
CSeq: 26662 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 20.5.0
Content-Type: application/sdp
Content-Length: 239
v=0
o=- 29623145 29623145 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 11804 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
-- Called PJSIP/201
<--- Received SIP response (355 bytes) from UDP:201.202.203.204:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 101.102.103.104:5060;rport=5060;received=101.102.103.104;branch=z9hG4bKPj92692640-20ec-4fac-a8e8-44cc006be541
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204:5060;ob>
CSeq: 26662 INVITE
Content-Length: 0
<--- Received SIP response (546 bytes) from UDP:201.202.203.204:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 101.102.103.104:5060;rport=5060;received=101.102.103.104;branch=z9hG4bKPj92692640-20ec-4fac-a8e8-44cc006be541
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204:5060;ob>;tag=l-MAh0nazgobuWkhevsjuQZJ-2PFb.Nh
CSeq: 26662 INVITE
Contact: "User One" <sip:201@201.202.203.204:5060;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
-- PJSIP/201-00000003 is ringing
<--- Transmitting SIP response (559 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Contact: <sip:101.102.103.104:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0
<--- Received SIP response (936 bytes) from UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 101.102.103.104:5060;rport=5060;received=101.102.103.104;branch=z9hG4bKPj92692640-20ec-4fac-a8e8-44cc006be541
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204:5060;ob>;tag=l-MAh0nazgobuWkhevsjuQZJ-2PFb.Nh
CSeq: 26662 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "User One" <sip:201@201.202.203.204:5060;ob>
Supported: replaces, 100rel, norefersub
Content-Type: application/sdp
Content-Length: 320
v=0
o=- 3910135817 3910135818 IN IP4 201.202.203.204
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
c=IN IP4 201.202.203.204
b=TIAS:96000
a=rtcp:4007 IN IP4 192.168.1.4
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ssrc:44516472 cname:6999942761cdc29a
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
> 0x7f3ca0040020 -- Strict RTP learning after remote address set to: 201.202.203.204:4006
<--- Transmitting SIP request (430 bytes) to UDP:201.202.203.204:5060 --->
ACK sip:201@201.202.203.204:5060;ob SIP/2.0
Via: SIP/2.0/UDP 101.102.103.104:5060;rport;branch=z9hG4bKPj139c6422-3535-43e9-ba8d-b59f25eee856
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204;ob>;tag=l-MAh0nazgobuWkhevsjuQZJ-2PFb.Nh
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
CSeq: 26662 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0
-- PJSIP/201-00000003 answered PJSIP/200-00000002
> 0x7f3ca00ca840 -- Strict RTP learning after remote address set to: 201.202.203.204:4002
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- Channel PJSIP/201-00000003 joined 'simple_bridge' basic-bridge <fc2a3d2a-781b-4f2d-b29c-46d0e0d44d95>
-- Channel PJSIP/200-00000002 joined 'simple_bridge' basic-bridge <fc2a3d2a-781b-4f2d-b29c-46d0e0d44d95>
> Bridge fc2a3d2a-781b-4f2d-b29c-46d0e0d44d95: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'PJSIP/200-00000002' and 'PJSIP/201-00000003' in stack
> 0x7f3ca0040020 -- Strict RTP switching to RTP target address 201.202.203.204:4006 as source
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
> 0x7f3ca0040020 -- Strict RTP learning complete - Locking on source address 201.202.203.204:4006
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP request (549 bytes) from UDP:201.202.203.204:5060 --->
REGISTER sip:101.102.103.104 SIP/2.0
Via: SIP/2.0/UDP 201.202.203.204:5060;rport;branch=z9hG4bKPjhaF3zhiiYwkDhh5OQWD.O8dW49YT.pi9
Max-Forwards: 70
From: "UserTest" <sip:200@101.102.103.104>;tag=ONXLSpGhHd0rlH4H85QBRWfkZhWAsdUI
To: "UserTest" <sip:200@101.102.103.104>
Call-ID: 4BVBM8EXD36kkJn6yJxL0jtNoWDiJAqY
CSeq: 9387 REGISTER
User-Agent: Telephone 1.6
Contact: "UserTest" <sip:200@201.202.203.204:5060;ob>
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<--- Transmitting SIP response (578 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjhaF3zhiiYwkDhh5OQWD.O8dW49YT.pi9
Call-ID: 4BVBM8EXD36kkJn6yJxL0jtNoWDiJAqY
From: "UserTest" <sip:200@101.102.103.104>;tag=ONXLSpGhHd0rlH4H85QBRWfkZhWAsdUI
To: "UserTest" <sip:200@101.102.103.104>;tag=z9hG4bKPjhaF3zhiiYwkDhh5OQWD.O8dW49YT.pi9
CSeq: 9387 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1701147046/985c6358d1a6b6f2212d60096049ee27",opaque="5736a170355b5eea",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length: 0
<--- Received SIP request (838 bytes) from UDP:201.202.203.204:5060 --->
REGISTER sip:101.102.103.104 SIP/2.0
Via: SIP/2.0/UDP 201.202.203.204:5060;rport;branch=z9hG4bKPjN2BKdI7CN.-b0ZlCMompr23kcTOVyaHf
Max-Forwards: 70
From: "UserTest" <sip:200@101.102.103.104>;tag=ONXLSpGhHd0rlH4H85QBRWfkZhWAsdUI
To: "UserTest" <sip:200@101.102.103.104>
Call-ID: 4BVBM8EXD36kkJn6yJxL0jtNoWDiJAqY
CSeq: 9388 REGISTER
User-Agent: Telephone 1.6
Contact: "UserTest" <sip:200@201.202.203.204:5060;ob>
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="200", realm="asterisk", nonce="1701147046/985c6358d1a6b6f2212d60096049ee27", uri="sip:101.102.103.104", response="778ee921c79ac2902a008f896cdebf4b", algorithm=MD5, cnonce="NVuKMUUyStr7XLbfbnBjR.B0Gs.m74YK", opaque="5736a170355b5eea", qop=auth, nc=00000001
Content-Length: 0
<--- Transmitting SIP response (527 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjN2BKdI7CN.-b0ZlCMompr23kcTOVyaHf
Call-ID: 4BVBM8EXD36kkJn6yJxL0jtNoWDiJAqY
From: "UserTest" <sip:200@101.102.103.104>;tag=ONXLSpGhHd0rlH4H85QBRWfkZhWAsdUI
To: "UserTest" <sip:200@101.102.103.104>;tag=z9hG4bKPjN2BKdI7CN.-b0ZlCMompr23kcTOVyaHf
CSeq: 9388 REGISTER
Date: Tue, 28 Nov 2023 04:50:46 GMT
Contact: <sip:200@201.202.203.204:5060;ob>;expires=59
Expires: 60
Server: Asterisk PBX 20.5.0
Content-Length: 0
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP request (553 bytes) from UDP:201.202.203.204:5060 --->
REGISTER sip:101.102.103.104 SIP/2.0
Via: SIP/2.0/UDP 201.202.203.204:5060;rport;branch=z9hG4bKPjlmNKrO4.HyDyQY2m-Fy04nrSt4fH0gMe
Max-Forwards: 70
From: "User One" <sip:201@101.102.103.104>;tag=Hypu9g3dFiryavFXlyK.wK2hbj07QIIp
To: "User One" <sip:201@101.102.103.104>
Call-ID: 7M-06FxEoto1v3R2K0AkmgP5EfFj2h7i
CSeq: 49062 REGISTER
User-Agent: Telephone 1.6
Contact: "User One" <sip:201@201.202.203.204:5060;ob>
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<--- Transmitting SIP response (581 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjlmNKrO4.HyDyQY2m-Fy04nrSt4fH0gMe
Call-ID: 7M-06FxEoto1v3R2K0AkmgP5EfFj2h7i
From: "User One" <sip:201@101.102.103.104>;tag=Hypu9g3dFiryavFXlyK.wK2hbj07QIIp
To: "User One" <sip:201@101.102.103.104>;tag=z9hG4bKPjlmNKrO4.HyDyQY2m-Fy04nrSt4fH0gMe
CSeq: 49062 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1701147048/f5d14007150bfbe5d23612f6518f45d8",opaque="5e71bb232c7e06d8",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length: 0
<--- Received SIP request (842 bytes) from UDP:201.202.203.204:5060 --->
REGISTER sip:101.102.103.104 SIP/2.0
Via: SIP/2.0/UDP 201.202.203.204:5060;rport;branch=z9hG4bKPjLR7s5oFghInideh8BZJ2Tks7I5TmyLHL
Max-Forwards: 70
From: "User One" <sip:201@101.102.103.104>;tag=Hypu9g3dFiryavFXlyK.wK2hbj07QIIp
To: "User One" <sip:201@101.102.103.104>
Call-ID: 7M-06FxEoto1v3R2K0AkmgP5EfFj2h7i
CSeq: 49063 REGISTER
User-Agent: Telephone 1.6
Contact: "User One" <sip:201@201.202.203.204:5060;ob>
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="201", realm="asterisk", nonce="1701147048/f5d14007150bfbe5d23612f6518f45d8", uri="sip:101.102.103.104", response="2473c46d929c72e7c62e96a26e6ffb29", algorithm=MD5, cnonce="i7fetChIVMDPxbrphddTODn9GyZrUoDY", opaque="5e71bb232c7e06d8", qop=auth, nc=00000001
Content-Length: 0
<--- Transmitting SIP response (530 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjLR7s5oFghInideh8BZJ2Tks7I5TmyLHL
Call-ID: 7M-06FxEoto1v3R2K0AkmgP5EfFj2h7i
From: "User One" <sip:201@101.102.103.104>;tag=Hypu9g3dFiryavFXlyK.wK2hbj07QIIp
To: "User One" <sip:201@101.102.103.104>;tag=z9hG4bKPjLR7s5oFghInideh8BZJ2Tks7I5TmyLHL
CSeq: 49063 REGISTER
Date: Tue, 28 Nov 2023 04:50:48 GMT
Contact: <sip:201@201.202.203.204:5060;ob>;expires=59
Expires: 60
Server: Asterisk PBX 20.5.0
Content-Length: 0
<--- Transmitting SIP response (879 bytes) to UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.202.203.204:5060;rport=5060;received=201.202.203.204;branch=z9hG4bKPjg8pfEn5kAfDfizIqdcqg2zK--DG5LvBR
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
To: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
CSeq: 24994 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:101.102.103.104:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 3910135785 3910135787 IN IP4 101.102.103.104
s=Asterisk
c=IN IP4 101.102.103.104
t=0 0
m=audio 15850 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP request (428 bytes) to UDP:201.202.203.204:5060 --->
BYE sip:200@201.202.203.204:5060;ob SIP/2.0
Via: SIP/2.0/UDP 101.102.103.104:5060;rport;branch=z9hG4bKPjb4d06f46-55ef-4445-be51-9d7956b82e71
From: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
To: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
CSeq: 12704 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0
<--- Received SIP response (378 bytes) from UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 101.102.103.104:5060;rport=5060;received=101.102.103.104;branch=z9hG4bKPjb4d06f46-55ef-4445-be51-9d7956b82e71
Call-ID: gDy2eie2tpALgrS4vcFdQZjW5X48FjNi
From: <sip:201@101.102.103.104>;tag=cbaa298e-a9e8-4104-a236-2d4d697c3351
To: "UserTest" <sip:200@101.102.103.104>;tag=kF2LFqHMfi33bwPAsd9HCOUdRjGEXwre
CSeq: 12704 BYE
Content-Length: 0
-- Channel PJSIP/200-00000002 left 'native_rtp' basic-bridge <fc2a3d2a-781b-4f2d-b29c-46d0e0d44d95>
== Spawn extension (internal, 201, 2) exited non-zero on 'PJSIP/200-00000002'
-- Channel PJSIP/201-00000003 left 'native_rtp' basic-bridge <fc2a3d2a-781b-4f2d-b29c-46d0e0d44d95>
<--- Transmitting SIP request (454 bytes) to UDP:201.202.203.204:5060 --->
BYE sip:201@201.202.203.204:5060;ob SIP/2.0
Via: SIP/2.0/UDP 101.102.103.104:5060;rport;branch=z9hG4bKPjc2e45960-1534-4fc8-bac9-54807f83bbfa
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204;ob>;tag=l-MAh0nazgobuWkhevsjuQZJ-2PFb.Nh
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
CSeq: 26663 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0
<--- Received SIP response (385 bytes) from UDP:201.202.203.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 101.102.103.104:5060;rport=5060;received=101.102.103.104;branch=z9hG4bKPjc2e45960-1534-4fc8-bac9-54807f83bbfa
Call-ID: 9f432c16-8951-4bb4-a7b8-3a0522469e5c
From: "UserTest" <sip:200@10.0.0.209>;tag=4b99399f-2b7e-48f6-bcbd-6c9aaf5e5dbd
To: <sip:201@201.202.203.204:5060;ob>;tag=l-MAh0nazgobuWkhevsjuQZJ-2PFb.Nh
CSeq: 26663 BYE
Content-Length: 0
Asterisk sent a 200 OK to an endpoint (200) to indicate answer. Asterisk never got an ACK to indicate it was received. The call was then hung up. This is always network or configuration in some way - behind NAT and port not forwarded, pjsip.conf not configured to know it is behind NAT, a firewall.
Thanks for the details.
Could you please help me on where to provide the mentioned configuration, to make this working.
I can’t. You have to diagnose it yourself further. Asterisk is sending a SIP request of 200 OK, it is either not being received by the remote side or the response from the remote side is not making it back to Asterisk. You therefore have to do packet captures and examine how traffic is flowing and try to figure out where it is getting lost, and then fix it if possible.
Thanks for the input.
[Assuming that we don’t have/missed any configuration specific to, handle this scenario.]
As mentioned, let me try to figure out where the packets are getting lost in the client side.
Does enabling tcp ( instead of udp) helps to solve this packet lost issue in client ?
Possibly? Or it’ll just mask the issue in that area, and it’ll manifest elsewhere eventually.
Thanks for the input.
If you’re behind NAT, you likely want to include:
external_media_address
external_signaling_address
To make sure the other end it told where to reply to. If that has anything to do with your problem?
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